Whirlpool

Hi all,

I am posting this thread as I just took receipt of the Siemens C470IP phone with 2 extra handsets. I plan to report my experiences with this phone on my home Trixbox(Asterisk) system as three individual extensions. For the WP community at large I'll also do some tests with it directly connecting to my VSPs: iiNet and Pennytel. I am happy to take questions and to try things out for you as well. Apologies for the long first post

OK lets start.

Information on the product can be found at www.gigaset.siemens.com

Why did I buy this phone

I wanted a cordless phone system for my home that takes advantage of the fact that I have an Asterisk (trixbox system). My current analogue cordless phone is 5 years old and is due for replacement and after diving into the VOIP world in a big way, I did not want to buy any new legacy telephony infrastructure.

I wanted each cordless phone to have its own SIP definition so that each phone can be treated as an individual extension, with one gateway/basetation. This would allow me to give each extension its own voicemail, support multiple simultaneous calls and so forth. This could have been achieved with an ATA, basetation and handset for each extension. With my Asterisk box I probably would have bought 3 FXS ports, instead of 3 ATAs but I would have still needed 3 base stations. That would have been very messy, bulky and would require a new big powerboard :).

A genuine IP aware phone would also allow me to try out some of those features not possible on analogue phones like centralised phone directores, email, presence notification, rss feeds etc.

What others phones did I consider

I looked into WiFi phones, but ultimately ruled them out due to battery life. WiFi was never designed to be used on a handheld device which have small batteries. Also I thought they were quite expensive, $400 per handset for the Linksys WIP330 although the Linksys WIP300 was a lot less at around $160 per handset.

Once convinced that I needed a DECT/SIP gateway device, I went looking around 6 months ago and there was not a lot. I looked closely at the DualPhone. But I was not impressed with its quality and support. It was originally call a LANPhone and support locally for it dropped off and was picked up later on by another distributor. It is a new manufacturer and all up I did not feel confident with the phone. Search on the WP forums for lanphone and/or dualphone and you will know what I mean.

Later I heard that Aastra was releasing exactly what I was looking for. They sold the base station/gateway (RFP32) and handsets (DECT142). Eventually I found out how much they were going to retail for, $1500 for the gateway and around $400-$500 for each handset. This was way too much for my home system. It is designed for office environments so that you can setup DECT based cells that can hand off the handset to different base stations when people roam around the office. They even have outdoor variants of the RFP32.

I gave up a for a while an looked again earlier this year. DLink have released the DPH-C160S. I actually encountered this phone last year when scouring the Internet, from the manufacturer in China. After asking some questions on features, manuals etc I thought that it will be very buggy, and the support will simply not be there. The quality did not seem the best either. Now that D-Link is selling it things might be better but I was still wary and besides they don't sell additional handsets.

The only serious contender was the Snom M3, due to be relased next month. From all reports it will be do exactly what I want but at around $400 compared to the Siemens, I could not see what the extra money would have got me.

Other things that I considered. The Siemens Gigaset brand is a well known and established brand. They have a full range including older models, models with answering machines, and high end High Definition Sound models (why that would be useful for telephone quality audio, I don't know, which I did not buy that one). This means support, including community support, will be there, quality will be better and the firmware/software much more stable. For some reason it just hasn't reached Australia in any big way

Where did I buy

I looked for the phone in Australia and the best price I could find was $270, delivered from ryda.com.au. Later I found that Freshtel are selling them as well for around $220, delivered. At that price I thought they were locked, but a Freshtel rep confirmed the first 500 will be unlocked, but subsequent batches will be locked see here forum-replies.cfm?t=920389#r17. The other thing is that for some unknown reason they are not selling additional handsets in Australia, the C47H. I am sure any GAP compatible DECT handset would be able to connect, but I wanted the full set of features, including IP aware features. I tried calling Siemens locally but nobody called me back. Interestingly if you look at the Siemens website you can download an Australianised manual for the C470IP but not the C47H.

As soon as I realised additional handset were not available, at least in the short term, I went overseas. I've bought heaps of stuff from the US and HK, and a small amount from the UK so I am comfortable with doing this if the price is right. I ended buying it from Germany www.voipdistri.com. The delivered price for my C470IP plus 2 extra C47H handsets was $383. A previous quote for only one extra handset was $278. Interpolating from that, the standard unit with a single hand set would be around $170-180. So I definitely saved money and got my additional handsets. Exchange rates has not varied that much so these prices are still reasonably accurate.

I was waiting quite anxiously for them as it took a while. I paid via paypal on the night of 31/1/2008, it was posted on 4/2/2008 and I received it today. It was sent via DHL, so I was expecting it to arrive much more quickly being a courier, but as I later found out there are lots of DHL companies, and mine was sent via the equivalent to Australia Post. Mathais was very helpful through the whole process. After giving me the DHL tracking ID, he later got me the match code, that is the number that Australia Post understands. For a long time the web based tracking systems, www.dhl.de and www.auspost.com.au, reported dispatched from Frankfurt on 11 February 2008 at 1:14am enroute to final destination. On Monday it totally disappeared from the Australia Post system and after ringing them I found out it is because it has cleared customs and is "in the mail". Now that I know what is meant to happen, the leadtimes, and to not expect real time information on the tracking system, I will not be so anxious next time. If anybody wants to order this way, just whim me and I'll give you his email address, I definitely would consider using them again. You could always ask for a more expensive courier as an option as well.

Note when buying this from overseas you will get Euro AC adaptors. The voltage is right so all you need is a cheap plug converter. I bought el-cheapo ones from ebay $13.50 for 4, delivered. They do the job and the AC adaptors are so light that the extra few centimetres length from the converters do not result in them coming out of the socket. I'll see how these go but the worst case is that I spend a little more on better quality ones. Replacing the AC adapter is not an option due to the non standard power connector on the charging cradle, unless you want to cut and resolder wires. The base station uses a standard connector but it is right angled to keep the cable neat. As far as I know replacement ones from Jaycar etc come with an assortment of power pins and are not right angled. If you replace the power pack the only down side is the cable won't be as neat at the back. At the end of the day cheap plug converters was what I went with.

This device can also connect to the PSTN so it comes with a German telephone cable so you need to replace that with a standard RJ11-RJ11 cable.

One last thing when you turn it on, it is all in German, so you have to navigate through the menus to change the language to English, not particularly easy for one who knows no German but I worked it out in the end with a bit translation help via Google.

Of course when buying anything from overseas you have to take several factors into consideration: delivery lead times, little chance of any warranty being honoured locally, need for plug converters, possibility of issues with Customs (no A-tick on these phones even though they are identical to the local product). Although a possibility, I've never had to pay GST on anything bought from overseas including these phones. I don't think anybody will if they just buy for themselves, rather than commercial quantities.

After charging and changing the language on the handsets to English, all I have done so far is to register all the handsets to the base station. The real interesting stuff comes up in the next few days. Stay tuned.

Z

Firmware Update

As it turns out this is incredibly easy and painless. All you do is connect to the base station via a browser and click the Update Firmware button in Settings --> Miscellaneous.

It checks for a firmware update and downloads and updates it if required. The only disconcerting thing was that there was no indication on how long it takes. During the Firmware update the base station is off the network and the handsets are trying to find their base station by flashing Base 1 on the handset screen. I didn't time it but I think it took at least 5 minutes. You know it is complete when the handsets successfully re-register with the base station with their screen display showing INT 1, in the case of the first handset.

Z

Keep it coming.

Reading with interest.

Ian

Connecting to Asterisk

Just a reminder that my Asterisk system is a Trixbox 2.4 distro.

All I did in Asterisk was define 3 more SIP extensions, intending them to correspond to the 3 handsets. In the Siemens C470IP all I did was connect via a web browser and went to the Settings -> Telephony -> Connection section. In that section there was space to enter 6 SIP providers. There was also one fixed configuration for Gigaset.net, which I disabled and might experiment with later on. All I had to do was to define 3 SIP providers being the three new extensions that I just configured in Asterisk.

It was really quite straight forward. The only thing different to what I've seen in others is that it has settings for domain, proxy server, registrar server and outbound proxy. I had to set them all to the address of my Asterisk system.

The interesting thing in the C470IP is the way they treat the SIP providers. Each provider, including the PSTN, can be accessed from any handset by adding a suffix at the end of the dialed number, #2 for example will force it use the 2nd SIP provider. This is not quite what I want as I want each handset to be tied to only one SIP provider, being the extension in Asterisk. To do this you need to go into the Settings -> Telephony --> Number Assignment section. Here each handset can select one of the SIP providers for out going calls, and any number of the SIP providers for incoming calls plus the PSTN and Gigaset. This means each handset must choose one and only default provider to make outgoing calls, but can have any number of providers for incoming calls (from none to all), so that the handset rings when any of those providers gets an incoming call. Apparently all this can be overridden if there was dialplan as well. I will test all this at later when I test with direct connections to the VSPs without the Asterisk box.

So In my case I would select handset 1 to have SIP provder 1 as the outgoing, and only one selection for the incoming, being SIP provider 1. Similarly, I would do the same for the other 2 handsets. It's all very easy to do on the Web GUI with radio buttons and checkboxes.

Z

Message Waiting Indication (MWI) - Asterisk

This is the first feature that I found that does not seem to be working properly.

I have set up voicemail accounts for the extensions so when a voicemail is deposited, MWI should be sent to the phone. If I was using the x-lite softphone it would be a little envelope icon in the display screen.

For the C470IP it shows an icon of a reel-to-reel tape along with the number of voice mail messages, which is quite neat. Additionally it flashes a bright red LED behind a couple of the buttons so that you can see there is voicemail when the phone is not in your hand, for example on the charging cradle.

The problem with the C470IP, is that it only works once, after you have rebooted the base station. MWI does not work for any subsequent voicemails. A scan of the internet reveals others reporting the same issue on this and other models.

I hope they fix it soon with a new firmware as it is a feature I would like. However, I can put up with it for a while, because before being diverted to voicemail it rings the extension first. When that happens I get missed call information, just like a mobile phone, and the same LED starts flashing, so that I know somebody tried to call and there might be voicemail, even though when looking at the screen there is no MWI.

I don't know if the same issue will arise if I connect directly to the VSPs (iiNet and Pennytel in my case). I've never used the VSP's voicemail so I have to check to see if they support MWI anyway. Another thing for me to test when I get to testing direct connections to the VSPs.

Z

Excellent thread and information, thanks for going to all the trouble!

Based on the information you have provided so far, I downloaded the manual and was pleased to see the overall feature-set and VoIP call controls provided (ie, dial plans etc).

But given the manual's overall lack of information about "fixed line" functionality, I am presuming that there is no call forwarding (via VoIP) or answering machine functionality available for incoming "fixed line" calls. The manual does at least state that incoming "fixed line" calls do not show as "call waiting" when on a VoIP call, which would appear to support this assumption. If this is correct, is would appear to me to be the only apparent weakness in an otherwise very functional system.

Can you confirm?

zforum69 writes...

One last thing when you turn it on, it is all in German, so you have to navigate through the menus to change the language to English, not particularly easy for one who knows no German but I worked it out in the end with a bit translation help via Google.

Hi Could you please confirm if the same can be done for the answering machine? say for the C475 or S675 models?
I believe the answering machine language can not be changed. Does anyone know if this can be updated via a firmware update?
I can get these phones from Sweden but the answering machine is not in english.
Cheers

C475IP has answering machine function, C470IP does not.

However if sourced from Germany the language for answering machine pre recorded announcements and voice prompts would be German. Certain C475IP versions may have other languages such as Italian and French as additional choices. There is an international version made with Czech, Slovene, Croatian and English choices. Not sure which particular version of C475IP is sold in Australia, because I could not find an Australian manual for it. As far as I can see only www.telephonesonline.com.au is selling it here for $328.90. One would hope that it has English as a choice. :)

For HDSP (G.722) models, development of an English language version of the answering machine function has been long delayed for the S675IP and it is now possible that English will only be available in the newer S685IP with Bluetooth. Not certain if this information is accurate though, it was just something I read in a forum somewhere.

Edit: I do not think the answering machine language choices are able to be changed by firmware. It may be possible but I have not read of anyone doing it. For example there is a Scandinavian/Nordic version of S675IP which has Swedish, Danish, Norwegian and Finnish as choices.

Markovich writes...

But given the manual's overall lack of information about "fixed line" functionality, I am presuming that there is no call forwarding (via VoIP) or answering machine functionality available for incoming "fixed line" calls. The manual does at least state that incoming "fixed line" calls do not show as "call waiting" when on a VoIP call, which would appear to support this assumption. If this is correct, is would appear to me to be the only apparent weakness in an otherwise very functional system.

Can you confirm?

Happy to try something out for you. I'm just a little confused what you are asking me to do. You refer to an answering machne, but I have the C470IP without an integrated answering machine. The C475IP (not sold in Australia) is the one with the integrated answering machine.

I think you are asking me to test/compare diversion of an incoming VOIP line to an incoming PSTN line.

When you have turned on diversion on an incoming VOIP line the diversion is like a mobile phone diversion. You can set a number for it to divert to. However unlike a mobile phone the choice to divert on all calls, no answer or busy is mutually exclusive. For example if you have set call diversion on busy, it will not divert on no answer.

Diversion on the PSTN works differently, it seems to dial a special set of characters, hoping the carrier sets the diversion. For example if you set all calls to divert to phone number 12345678, and you press send it dials *21*12345678#. I haven't seen anywhere where I can set the special dial prefixes/suffixes for PSTN diversion codes, if those are incorrect for our carriers.

I hope that answers your question. If not post back a clarification and I'll try again for you.

In my situation none of this is really relevant, as I have an asterisk box that terminates my PSTN line. I would not normally have the PSTN line connected on the Siemens. A call on my PSTN line goes through Asterisk and then via SIP to the Siemens. I still want to think about how I might use that PSTN connection, I've got a few ideas but that will be for another installment.

Z

jag08 writes...

Hi Could you please confirm if the same can be done for the answering machine? say for the C475 or S675 models?

I have the C470IP, and as per kasi's post this one does not have an answering machine.

Z

DTMF

The C470IP allows you to select any number of the 3 modes for DTMF over VOIP (audio, RFC2833 and SIP Info). These are needed in IVRs that request you to press your keypad according to menu options.

I found that RFC2833 works best. My testing revealed, nothing works with audio, Asterisk works with SIP Info but my calls through my VSPs, iinet and Pennytel, do not recognise it. RFC2833 is the one that is recognised by all.

Z

Simlutaneous phone calls

I have always known the C470IP can only support two simultaneous VOIP calls plus the PSTN. I never thought this would be a limitation as I could not conceive in a home environment when I ever needed to do 3 simultaneous voice calls. If I really wanted to do 3, my plan was to connect the C470IP's PSTN line to an FXS port on my Asterisk box. That would give me the ability to do 3 simulateous SIP VOIP connections through the Astersisk box, but 2 via SIP to the C470IP handsets and 1 via analogue PSTN to the C470IP handsets.

When trying to work out how I want my home system to work I thought a good start was to ring all handsets when a call comes in. The logical thing to do in Asterisk is to setup a ring group to ring all extensions when a call comes in. To my initial surprise only two of the extensions rang. I quickly realised this is probably due to the 2 SIP call limitation, even though I have not got an active call going. All three extensions were registered with Asterisk, however when Asterisk tried to call each extension in the ringgroup, the Asterisk CLI reports only 2 of them successfully ringing.

This is not a big issue for me as I'm sure I can work out an alternative. For example I could ring a hunt group that tries each extension in turn, or I could get the C470IP to ring two or three extensions when an incoming call from one of the providers or the PSTN comes in. But it is something that you should keep in mind.

Z

zforum69 writes...

The C475IP (not sold in Australia) is the one with the integrated answering machine.

Ah, thanks. My curiosity is coming from a slightly different angle because I don't have, or intend to use, an Asterisk box or similar, so was looking for any SPA-3102-style functionality where the incoming "fixed line" (FXO) can be passed off to VoIP (when on a VoIP call, an incoming PSTN call can be forwarded to another SIP extension or VoXaLot voicemail etc).

I can see this isn't the case, and it also sounds like the 475IP is using a fixed-line/answering machine in parallel to it's VoIP/mailbox functionality, rather than integrating the two. Still sounds like a great phone, so I'm curious as to whether we'll see the 475IP here, as well as multi-handset packages.

IP Features - Info Services

I thought I'd give some of the IP features a go, this time its Info Services.

When logging into the C470IP you browse to the Settings --> Service page. There you can select to turn on Info services. The way you configure it is via an external Gigaset web portal. The same config page gives you your userid and random looking password along with a hyperlink to the portal. After logging into that portal you essentially have two choices: weather or RSS feeds.

If you choose the weather you can select world selection, which will give you the weather of some random well known city in the world. Alternatively you can type in a city for it to search for. If it is in the database it comes up as an available selection. For me it found Canberra and obviously I chose that.

To display the information on the handset you need to turn on the screen saver. There you have a choice of 1 of 3 pictures and clock. To display the information you need to set it to clock. If Info services was disabled a big digital clock simply gets displayed. If Info services is enabled, the clock is dislayed for 2-3 seconds then the weather. It shows the weather for today and tomorrow, minimum and maximum temperatures and a precis, e.g. sunny.

I found the RSS feeds very lacking. Yes they worked but it only shows the first line. I can't even find an RSS feed that has some reasonably informative one line information. This function is effectively useless because of it.

If you want to use Info services, just stick with the weather. Maybe a new firmware will give better RSS support.

Z

Markovich writes...

I'm curious as to whether we'll see the 475IP here, as well as multi-handset packages.

I'm after the 475IP. I emailed Siemens for information but they didn't reply.

You can get it in the UK from www.home-phones.co.uk/as...or=any&­ PT_ID=all

Available mid-March.

Ian

Iangh writes...

I'm after the 475IP.

Perhaps you could enquire about current availability from here:

www.telephonesonline.com...y_price_SIP.html

Thanks for the review of the C470IP. How is the sound quality and have you tested direct connection to your two VSPs yet?

kasi writes...

Perhaps you could enquire about current availability from here:

www.telephonesonline.com...y_price_SIP.html


That's very interesting, maybe they are available in Australia, but I'd be somewhat wary as the gigaset web site does not have an Australianised manual, as mentioned by kasi in an earlier post.

At $328 plus delivery that's quite a bit more than what I could get it for from overseas, should be very close to the $200 delivered. That's the sort of price difference that would make me buy overseas without question, except for kasi's earlier post about voice prompts possibly not being in available in english. Due to the cost savings if I was in the market for one I would purse it with voipdistri.de and ask

kasi writes...

How is the sound quality

It sounds fine but I'm not sure how to make a judgement more specific than that. What would be the testing methodology. What do I compare to?

have you tested direct connection to your two VSPs yet?

Not yet but I will and will post here as promised. I suspect there are a few of you who are interested in that.

Z

Implications of base station rather than handset owning all the SIP registrations

In one of my earlier posts I mentioned how the handsets are tied to the base station, and what I did to map each Asterisk SIP extension to each handset. The point to note is that all the SIP definitions for the Asterisk extensions are with the base station, and then the base station maps it to the handset using DECT. The implication of this is that even if all the phones have been turned off, Asterisk still reports them as being registered, since the base station is still on.

This really is not an issue but may be a source of confusion in deployments with Asterisk. When you use things like HUDlite to get an operator's view of the status of all the extensions and trunks, all the C470IP extensions will always be reported as active. This is a little different if you use a softphone as an extension. If the softphone is not running, HUDlte reports it inactive.

Looking at the CLI in Asterisk when a call comes to a the handest that is turned off, the SIP registration is still active but Asterisk will receive a SIP response 480 "Temporarily not available" from the C470IP prompting my Asterisk box to play message a that the person on extesnion so and so is unavailable and then onto voicemail.

It may also have implications if directly connected to a VSP on how exactly it will treat the C470IP being registered but the handset being turned off. I'll remember to examine this when I do the direct VSP connection tests.

Z

zforum69 writes...

It sounds fine but I'm not sure how to make a judgement more specific than that. What would be the testing methodology. What do I compare to?

Well I thought that you would compare it to what you were using previously, your previous cordless phone you mentioned in the first post. I don't know anything about a Trixbox, but I assumed it is a system based on card in a computer and functioned in a similar way to an ATA, but with extra features. I understand that if you haven't used an ordinary ATA together with a cordless phone before then you can't compare the sound quality with an integrated DECT/SIP phone. I was just interested because I read of one user who reported that the sound quality was better with his Siemens DECT/SIP phone, than with a separate ATA and cordless phone.

"It sounds fine" is all I really wanted to know.

Not yet but I will and will post here as promised.

No worries, whenever you get to it, I just wasn't sure whether you had already done this and had been reporting on other functions first.

kasi writes...

I was just interested because I read of one user who reported that the sound quality was better with his Siemens DECT/SIP phone, than with a separate ATA and cordless phone.

It's very possible. It is one of the reasons why I decided to stay away from the ATA route and go straight to a TDM400 PCI card on a dedicated asterisk box. I would not be surprised if my original non-C470IP setup would sound better than a setup with an ordinary ATA. I might borrow somebody's ATA and perform my own subjective tests some time: cordless analogue phone, cordless analogue phone with ATA, cordless analogue phone with TDM400 card, and C470IP.

This little VOIP project of mine just gets bigger and bigger and never looks like finishing :).

Z

kasi writes...

Perhaps you could enquire about current availability from here:

www.telephonesonline.com...y_price_SIP.html


Yes, I noted that site in my email to Siemens Australia and also noted the price difference with the UK site.

I don't like being ripped off.

Ian

Online directories

Siemens Gigaset have setup a gigaset.net voip network that all its VOIP hardware can participate in. That is if two Gigaset IP phones have registered to the network they can make free calls to each other for example. Consequently you can also access the online directory for Gigaset.

The C470IP also has the abiltiy to connect to other online directories. You configure this via the web interface by Settings --> Services and there is a section for Online Directory. My unit came from Germany and I can configure the unit to use one of 5 different directories, all Eupropean countries unfortunately. I can also select to display the caller's name so that when somebody calls it brings up the caller's name from the online directory rather than just the number. When reading the Australianised manual it doesn't even mention it, so I assume it may have been disabled in the Australian version. Once you enable one of these directories it becomes accessible via the handset.

To access the online directories all you have to do is press and hold the address book key. If I selected klickTel, as my online directory provider, I will get three directories to search: kT phonebook, kT Yellowpages and gigaset.net. I presume the Australian version it will be like if the online directory was disabled, so it would skip the menu selection and go straight to the gigaset.net search.

The concept is very good and is an excellent feature, however for people here in Australia it will have no value as there is no online directory for Australia and the number of people on gigaset.net that might call you,or you might want to call will be some where close to zero. I have one of these phone and can't even be bothered registering.

For us think this feature is more like a teaser of what is possible. I would imagine later generation phones would use LDAP or web services query online telephone directories.

Z

Phone Directory

Via the browser you can upload your phone directory to an individual handset resulting in each handset being able to have its own customised directory. The file that it needs needs to be in vcard format. This all sounds real good. The manual even suggests that one can export ones Outlook contacts in vcard format and upload it. Although the claim is true it is not quite the same level of sophistication and functionality as if you were to do soemthing similar with a modern mobile phone.

First of all, no utilities have been provided to export Outlook contacts to vcard. Contrary to the manual's suggestion you cannot export Outlook contacts to .vcf format without a third party utility. I think you can do one at a time, but that would be useless, as you need to select all your contacts, and export them all to one .vcf file.

The manual shows an example of a vard format file with the person's name, three associated numbers, birthdays and an email address as well. The manual suggests it can read all this and ignore's all other fields. It's wrong. The only relevant fields are name and numbers. Furthermore, it dosen't store the 3 numbers against a single name. It will import them and have three entries of the same name to store the three numbers. There is no intelligence in the name processing. Typically the name is stored like N:Surname;Firstname in the vcard file, but the C470IP will store the name exactly in the same format as Surname;Firstname, with the semi colon. Not as straight forward as I would have hoped.

What I'm going to do is to export my contacts to a comma delimited text file, with only the name and numbers, and then write a script to write the vcard file suitable for the C470IP. Each number will have its own name entry, with a suffix at the end of the name eg: N:Firstname Lastname - M for the mobile phone entry. This should not be too hard to do.

I guess the good thing about all this is that you can store all your phone contacts on your PC and upload them all to your handsets. This has been a problem with multi-handset cordless phones, having to manually enter in names an numbers into each handset usign a numeric keypad. I was told some newer cordless phones have some sort of ability to connect to the PC to do just this as well

Z

zforum69 writes...

Asterisk works with SIP Info but my calls through my VSPs, iinet and Pennytel, do not recognise it

Curious... are these for calls via the Asterisk box or direct from the handset? Do you allow re-invites for your Asterisk extensions?

I've got a similar setup with a Snom M3 and a Snom 320 both connected to an Asterisk server (using INFO for DTMF), which has trunks to iiNet and Pennytel (using RFC2833) and it all works fine.

You've got me thinking I should run a parallel thread to yours based on the M3 - but I've only got one handset, and might have trouble getting downtime for testing :-)

meric writes...

Curious... are these for calls via the Asterisk box or direct from the handset?

Via Asterisk. Test with direct connections are yet to come.

Do you allow re-invites for your Asterisk extensions?


No. This is the default when you add another SIP extension using Trixbox 2.4.2

You've got me thinking I should run a parallel thread to yours based on the M3

That would be good, I for one would be interested in it. As per my original post I was was seriously considering the Snom M3 but I could not justify the price difference.

Z

Direct connection to VSP - Pennytel

My setup at home uses an Asterisk box with the C470IP, but I realise most would not have such infrastrcutre. This is the first of my posts using the C470IP directly connecting to the VSP, which would be the configuration I'd expect most would use.

Before I go on, let me remind you all that this is a genuine IP phone, so it does not need an ATA. The base station simply needs to connect to Internet via its inbuilt Ethernet port. This is the sort of phone you would consider if you are in the market for a new cordless phone system, don't have any existing VOIP infrastructure other than Internet access, and want to utilise VOIP. You'd also consider this phone if you want the mutli-line capability that VOIP offers, instead of having an ATA and cordless phone with individual base station for every line. You might also want this phone if you want to consolidate some of your boxes by getting rid of your ATA.

First of all Pennytel does not seem to support MWI. I confirmed this by intially testing with the x-lite soft phone, so there is no testing on that.

Setting up the definition for Pennytel was quite straight forward. Similar to setting up an Asterisk extension you need to define the settings found under Advanced Settings. Put your 11-digit Pennytel phone number in the Authentication Name, Username and Display Name. Authentication password is obviously your Pennytel SIP password. Set the domain, proxy server, registrar server and outbound proxy to sip.pennytel.com.

I also had to rearrange the default codec arrangement for Pennytel to work, unlike Asterisk where the default was fine. In Settings --> Telephony --> Audio you can set the codecs. You can choose to optimise for high bandwidth, low bandwidth or your own preference. Select your Own preference. Further down the screen each SIP definition has a list of codecs you can use which can also be reordered. What I did was to move G.729 to the top. I also had to set Enable Annex B for codec G.729 to Yes. Note that this is a global setting. Once this was done it all seemed to work fine, inbound and outbound calls

When a call comes in you see it ring on your handset. Like a mobile phone, if you chose to reject it, it gets forwarded to Pennytel's voice mail if enabled. If the phone is turned off, it gets forwarded to Pennytel's voicemail as expected.

If you set your C470IP to ring two handsets on an incomng Pennytel call and you answer it on one, I found that you don't get a missed call on the other handset, which is exactly what you want.

Not much to say about making two simultaneous outbound calls other than is works fine.

An interesting thing is if I am on the phone and an inbound call comes in Pennytel will think I'm unavailable and divert to voicemail if enabled. However, if I enable call waiting then the other handset will ring, as well as hearing the call waiting pips on the active handset. The incoming call can be picked on the other handset while the active handset continues with their call, or the existing call put on hold on the active handset while the call waiting call is taken. 3-way conference is also available at this point as well. BTW the C470IP has music on hold which is the very basic chimes in some sort of tune, but at least the caller gets some sort of feedback that the line did not go dead.

One thing to note call waiting in the C470IP is a universal setting for all SIP accounts.

Z

IP Features - Time

Thought I'd quickly make mention of this as it is a nice feature that I almost overlooked.

Although when you first turn on your phone there is a prompt to set the time, the C470IP actually uses NTP. When you have it connected it to the network you just go to the relevant webpage, select your country, time zone, whether or not DST is observed and your done. You can overwrite the European NTP server that it has as default, but I left it as is.

The good thing about this is that the time will always be accurate, as it regularly resynchronises with the NTP server. According to the manual that is daily, restart of the basestation or when time settings have changed.

Z

Zf,

A very interesting thread on the C470IP.

I have the S450IP which is an earlier model with similar features.

I would appreciate your comments about the responsiveness of the phone in certain situations such as;

(a) Press the Hangup key - how long until the line actually hangs up. The S450IP seems to take 1 or 2 seconds from pressing the hangup key to actual disconnection.

(b) Time to resolve a name in the address book on an incoming call. Again on the S450IP, the number of an incoming call displays first, then in 1 to 2 seconds it resolves the name (assuming you have the phone address book populated)

(c) General responsiveness when navigating of the menus on the phone. With the S450IP I tend to overshoot the entry I want and have to go back because the display lags the pressing of keys.

You mention the Call Waiting Option, where is this located as I cannot find it on the S450IP web interface.

Trevor

TrevorB writes...

I would appreciate your comments about the responsiveness of the phone in certain situations

Shouldn't be a problem. I'll look into your questions when I get home tonight.

Z

TrevorB writes...

(a) Press the Hangup key - how long until the line actually hangs up. The S450IP seems to take 1 or 2 seconds from pressing the hangup key to actual disconnection.

Pretty much instantaneous, sub-second. While I was testing I was watching the Asterisk CLI to make sure hangup was occurring right when I pressed the button.

(b) Time to resolve a name in the address book on an incoming call. Again on the S450IP, the number of an incoming call displays first, then in 1 to 2 seconds it resolves the name (assuming you have the phone address book populated)

I don't have the address book populated so if there is effect on name resolution if the address book is near full I can't say, but otherwise no real delay.

(c) General responsiveness when navigating of the menus on the phone. With the S450IP I tend to overshoot the entry I want and have to go back because the display lags the pressing of keys.

Fine, however I did notice that some sub menus are slower. The first level menu is as quick as I can press with no possibility of the key presses being buffered. However the Sel. Service menu is slower and such that it does buffer my key presses if I press quickly enough, slow but for me not irritatingly slow.

You mention the Call Waiting Option, where is this located as I cannot find it on the S450IP web interface.

In the handset: Settings --> Sel. Services --> VoIP --> Call Waiting

Z

Hi Z,

Thanks for the test. It looks like the C470IP is faster than the S450IP for hangup and menus.

Re The Call waiting - I was looking on the Web interface, but I have now found it on the phone handset menu on the S45 handset.

You mentioned Call Waiting was universal for all SIP accounts, so I assumed the setting it would be on the Web interface. I will have to check out Call waiting on my system to see how it works as I don't use Voicemail on my PennyTel account.

If you want to test the Gigaset network, search the directory for "Trevor" and give me a call. I suspect the delay will be unbearable as the 1234#9 echo test had about a 2 second delay

Trevor

your reviews are tempting on the C470IP and am thinking of getting me one from Freshtel (thats if the unlocked batch is available).

However im pretty fussy about the appearance factor. Does the phone look appealing image wise :D

I mean i have the Uniden DSS7855 and i got it because it looked nice on my desk. Will this look nice too

broadbandfreak writes...

However im pretty fussy about the appearance factor. Does the phone look appealing image wise :D

I mean i have the Uniden DSS7855 and i got it because it looked nice on my desk. Will this look nice too

I don't know what the Uniden one you mentioned looks like and besides it is very much a subjective opinion so instead I've uploaded a hi-res image (2MB) of it here rapidshare.com/files/968.../C470IP.JPG.html. You can even see all the lint from the tissue I used when trying to clean it up for its modeling shoot :).

Z

zforum69 writes...

It sounds fine but I'm not sure how to make a judgement more specific than that. What would be the testing methodology. What do I compare to?

Using G729 I noticed that the sound quality is much better via Pennytel even on international calls to Philippines Mobile.
To get the same quality of sound with IINET (IITALK) I have to use G711 ulaw which unfortunately chews a lot data.
Hope someone can explain the reasons.

Iangh writes...

Yes, I noted that site in my email to Siemens Australia and also noted the price difference with the UK site.
I don't like being ripped off.

Another interesting site where additional hadsets are available at reasonable prices but no such offering for Australia!!!!!!! It seems we are destined to pay much more for whatever reasons!!!!!!!!
www.comms4you.co.uk/siem...-c475-p-987.html
In this case Siemens has not released any additional handsets by itself for Australia. Strange when they advertise that the base station can be used with 6 additional handsets!!!!!!!!
Any representation from Siemens on this site would be nice!!!!!!!!!

IP Features - Email

The is another one of the IP features inbuilt into the phone. Like other IP features, there is only one setting and is therefore a global setting for all handsets. I wasn't too concerned about Info Services being a global setting, however I would have liked an individual Email account to be tied each handset.

Setting it up is like everything else on the C470IP, all very straight forward. Just go to the relevant web page, put in your Email account name, password, POP3 server and then choose how often you want it to check (from 15mins to once per day). It only supports POP3, so you cannot use IMAP, POP/SSL or IMAP/SSL.

When there is new Email you hear a couple of beeps from all your handsets, and the message lamps start flashing (the same lamp used for missed calls and voicemail). The phone screens also show an icon of an envelope and a number indicating the number of unread messages. When you press the envelope key you will see the running total of voicemail, missed calls and emails, with anything that you have not looked at in bold. If you select Email you get a list of Emails showing who it is from (display name) and a date/time stamp. From there you can select From or Subject. If you select Subject it does the obvious and shows you the subject. If you select From it shows the display name again. I would have like it to show the Email address because I see no point in showing the display name twice.

One interesting thing, because the Email account is global, after you have viewed the Email list the message light stops flashing on all phones. In otherwords with Email, whatever you do with one handset reflected in all handsets, so you don't have to go to each handset to clear the message waiting lamp.

Z

Edit1: Deleting the Email from the handset, does not just delete it from the list, but actually deletes it from the POP3 server. When the email is eventually deleted or moved from the inbox the C470IP will reflect the new list on its next update. As a rule, since you can't read the email on the handset I'd expect one would not delete email from there, unless it is something like SPAM.

Edit2: This is regarding my reference above about selecting From on an email and only seeing the display name again. After reading the manual, apparently it is supposed to show the full email address. This clearly does not happen so it is a bug.

zforum69 writes...

In otherwords with Email, whatever you do with one handset reflected in all handsets, so you don't have to go to each handset to clear the message waiting lamp.

But you stil have to go to each email to delete it on a handset. A desiarable feature here would be a "Delete ALL" Feature.

jag08 writes...

But you stil have to go to each email to delete it on a handset. A desiarable feature here would be a "Delete ALL" Feature.

I personally don't think so. When you delete on the handset it deletes the email from the POP3 server, (this probably something I should have mentioned in my post). Normally one would not do this as you can't read the email from the handset. When you eventually read and delete the email or move it from the inbox, the C470IP will reflect that when it next updates.

Anyone got one of these babies doing MWI in any way shape or form? I just picked one up from freshtel, did a FW upgrade (swheet), but the NOTIFY messages get a SIP/2.0 481 Call Leg/Transaction Does Not Exist

I have a funny feeling that this is because my NOTIFY includes a Call-ID and the liboSIP that the gigaset uses probably sees this as an out-of-dialog thing and mushes it...

If someone does have this working a SIP trace would rock ;)

Peter Childs writes...

Anyone got one of these babies doing MWI in any way shape or form?

Some sort of shape I guess. Have a look at one of my earlier posts in this thread forum-replies.cfm?t=927308#r5.

Z

zforum69 writes...

Some sort of shape I guess. Have a look at one of my earlier posts in this thread forum-replies.cfm?t=927308#r5.


Hmm... I have a funny feeling Asterisk probably cache's the call-id from the first subscribe and doesn't update it, so when the event package/subscribe expires and the c740ip subscribes with a new call-id asterisk still sends notifiy with the old one... I'll run up an asterisk tomorrow and have a squiz.

What version of asterisk did you try it on?

Peter Childs writes...

What version of asterisk did you try it on?

Trixbox 2.4.2, so that uses Asterisk 1.4.17.

Z

zforum69 writes...

Trixbox 2.4.2, so that uses Asterisk 1.4.17.

Ok. So the Gigiset probably does the correct thing in terms of the RFC. Phones should 'SUBSCRIBE' for the message-summary Event, and in that subscription is a Call-ID which forms part of the 'dialog' for the transaction.

in sip.conf [general] set
subscribemwi=yes

NOTE THIS MAY BREAK PHONES THAT DON'T SUBSCRIBE that you send NOTIFY MWI to -- its a global setting... such is asterisk...

You also need to ensure that whatever the SUBSCRIBE is to exists in the context for the device. With things like trixbox that would probably be the case, but if you get 404 responses to your SUBSCRIBE you need to make a matching extension for the context...

ie in sip.conf if I have device
[470]
type=friend
host=dynamic
secret=470
nat=no
canreinvite=no
context=handset
mailbox=470

I need to have _something_ in extensions.conf due to a 'check' done in chan_sip...

[handset]
exten => 470,1,NoOp("Hit 470 in handset context" )

I started a stub on www.voip-info.org/wiki/v...s+Gigaset+C470IP if people want to add stuff specific to asterisk and this device.

That's great Peter. I'll have a look at it tonight.

BTW, being an Internode rep are you testing this for NodePhone?

Z

zforum69 writes...

BTW, being an Internode rep are you testing this for NodePhone?

Hmm... it was more of an interest thing. I liked the specs on the Snom M3 but my budget didn't reach that far. The MWI doesn't work with NodePhone as our platform doesn't support the SUBSCRIBE part of the MWI RFC (dolt) -- so I'm sure the wife is going to be beating me with a stick....

OK I gave it ago and cannot seem to improve things.

Previously I thought the only way to get MWI to work was to reboot the base station. In fact all I needed to was deactivate and reactive the SIP provider via the C470IP web interface.

When I do this and look in the SIP debug I see the subscribe, and the MWI lights up accordingly.

However, as before that is the last time it works. If I deposit an additional voicemail it does not update. I have to deactivate and reactive the SIP provider via the C470IP web interface to make it work again.

Peter Childs writes...

in sip.conf [general] set
subscribemwi=yes

This seemed to make no difference. The subscribe still worked after enabling the SIP provider and that was the last of it. I never did get 404 response to the subscribe message as the only time subscribe was done was on the activation of the SIP provider.

I do get the SIP/2.0 481 Call Leg/Transaction Does Not Exist that others have reported on the internet

Peter, how much testing did you do? Did you get it to update if another message was deposited?

Z

Message Waiting Indication (MWI) ... continued

While doing some more tests with MWI after Peter's suggestions above, I decided to give it a break after getting nowhere. I accidentally left some voicemails on the Asterisk server after clearing the MWI on the handset. Out of the blue while watching TV I got the beeps, the flashing message lamp, and the MWI on the handset with the number of voicemails. Luckily I left PUTTY on with SIP debug enabled, all logging to a text file. It seems after an hour it re-subscribes thus getting the MWI. The elapsed time is only an estimate as the screen log does not have timestamps against each line.

I confirmed this again this morning, by clearing MWI on the handset knowing full well there are still voicemails on the Asterisk server. Before going to work I asked my wife to call me when she notices the flashing message lamp and the MWI icon on the handset. She called me around 1.5hours later, but she could not say when it occurred as she was not near the phone around the 1hour mark.

On the C470IP, if you press the message button to see the number of voicemails and press OK it then dials the network mail box. This act is what clears the MWI on the handset, even if you hangup straight away without clearing the mailbox. The C470IP does not actively interrogate the network mailbox to determine if MWI needs to be changed or cleared. My latest experiences suggest it may, however, do this every 60mins.

Z

SUBSCRIBE sip:470@192.168.1.10 SIP/2.0.
(...)
Event: message-summary.
Expires: 3600.

So the phone will send a SUBSCRIBE every 60 mins, and asterisk will only send a NOTIFY when
. message count changes
. on receipt of a SUBSCRIBE....
. I _think_ from looking at mine it sends a NOTIFY after every REGISTER as well...

So you could try shortening your register timeout, or hacking the asterisk source code to be able to reduce the Expire time on the response to the SUBSCRIBE...

jag08 writes...

Using G729 I noticed that the sound quality is much better via Pennytel even on international calls to Philippines Mobile.

are you using a 3102 box? i would disagree with g729. today my g711 calls were much better than regular pstn.

Peter Childs writes...

So you could try shortening your register timeout

Tried all that and unfortunately that doesn't help making the SUBSCRIBE more frequent.

hacking the asterisk source code to be able to reduce the Expire time on the response to the SUBSCRIBE...

I'll give that a miss since I'm using using a trixbox.

The real fix is for the C470IP to correctly respond to the NOTIFY messages. In the meantime I'll put up with it. MWI will eventually show up within an hour, so I won't miss any messages, and the message lamp should light will light up anyway when there is a missed call before going to voicemail.

Z

kemikal writes...

are you using a 3102 box? i would disagree with g729. today my g711 calls were much better than regular pstn.

No I am using Siemens C470 ip phone.
My comment was in relation to comparison of calls between IITALK & PENNYTEL using the same codec G.729 .
G.729 is good quality on Pennytel compared to G.729 on IITALK. To get the resonable sound quality on IITALK I am having to use G711 ulaw. I don't know the reason why calls via Pennytel sound better even on lower codecs. Perhaps someone here can explain that.

IP Features - Instant Messaging

I have to admit I'm probably not the best person to evaluate this aspect of the phone. I never used chat programs or any other instant messaging application. Nevertheless I thought should test out the C470IP's support for this.

This is my first introduction into instant messaging so I only just found out what Jabber is, being the open standards based method for instant messaging. It's now got an official IETF name eXtensible Messaging and Presence Protocol (XMPP), and RFC's that define the standard.

The C470IP comes with a predefined Jabber account on the gigaset.net network. Since this is essentially a closed network without any transports to other instant messaging systems like MSN and Yahoo, I did not bother testing this. Instead I got myself a couple of accounts at jabber.org.au.

Firstly you still need a Jabber instant messaging client to set things up. I used Psi. You need this client to set up your "buddy list" being the users who you wish to chat with. I used my second test account as a buddy to this one which will be used on the C470IP. Once that is all done you can close Psi, as you are now going to use the C470IP as the new instant messenger client.

Like email there is only one account for everybody on the base station, rather than individual accounts for each handset. To configure the Jabber client just browse into the base station, choose Messaging --> Messenger. Then in Messenger Client select Other, and fill in the rest of the fields as appropriate.

To turn it on, go into the handset and choose Settings --> Messaging --> Messenger, where you will be prompted to start the messenger. That is about it.

From the phone you can see your other buddies and their status. The C470IP uses what looks like a pawn (i.e. the chess piece), as icons to represent the status of your buddies. Red means their off line, green means their online, and yellow means everything else (away, busy etc). From there you can chat or get a more detailed description of their status. From the phone you can also change your own status.

WRT the operation of the instant messenger client on the phone, I found no issues. The screen shows your status with an appropriately coloured pawn icon. If a message comes in, the message lamp will light up and flash. You then press the message key and select Messenger, and read the message. You can also reply to that message by typing a message using the numeric keypad.

I also managed to successfully get a chat going between the C470IP (via jabber.org.au) and a Yahoo client. Of course the Yahoo buddy and transport had to be initially defined and set up via Psi. The issue I found when using a transport to connect your Jabber account to other instant messaging systems like Yahoo, is that the user's status does not always propagate properly. That is definitely something to do with the the transport as the same issue exists if I use Psi.

In conclusion the instant messaging client seems to work fine. It obviously does not replace you PC based client as you still need it to set things up, but it seems like a convenient way to have a chat if you are not on your PC. I'm sure that generation who uses instant messaging and can converse at blistering speeds with various "TXT" abbreviations all from a numeric keypad would find this a useful thing to have. For me I'll stick with forums and emails using a PC with a full sized QWERTY keyboard :)

Z

All Siemens phones, including the C470IP, are on 30% off sale at Myer (at least in Melb city) at the moment.

That makes an unlocked C470IP $209.30

The only thing stopped me buying one, like many others, is the lack of spare handsets. I hope Siemens is reading this!

Timed Out writes...

The only thing stopped me buying one, like many others, is the lack of spare handsets. I hope Siemens is reading this!

Sometimes you have got to wonder about the marketing logic of some companies. It's not like they have to do anything special. The additional handset (C47H) is already in the C470IP package simply consisting of the AC adaptor, cradle and handset. There should be no regulatory issues as they would have been addressed with the C470, and they don't even have the excuse of having to source and supply an Australian AC adaptor.

Z

Open Source Code

I was just browsing the manual and noticed that the C470IP is based on open source code licenced under the GNU Lesser General Public License (LGPL). I had a browse on the gigaset web site and found the source code.

I had a friend of mine have a quick look over the code and he seems to think all is there. The website shows the latest source code for what they call the Baseline 97 which I assume is what the latest firmware 02097 is based one. It also implies the S450IP, C455IP, C470IP, C475IP, S675IP, CE450IP R and CE460IP R are all based on this common code base.

This is really good news. It means that the community would be able to code in new features and maybe fix some bugs as well. There have been several hardware items that were based on open source code that attacted active community support and were re-engineered to enhance its capabilities. Being under the LGPL not all of it is open source but I'm hopeful a substantial part of it is, thus allowing some community members who have development skills to take this product up, resulting in a vibrant and active development community for this phone.

Any takers? ..... You have at least one person here who is willing to be one of your testers.

Z

zforum69 writes...

I had a browse on the gigaset web site and found the source code.

I'd be surprised if it is all there. Snom do something similar for their IP handsets, but the published code is really only the Linux OS. The actual phone application is proprietary code and not released.

I had a friend of mine have a quick look over the code and he seems to think all is there.

Well... maybe it is. You'd want to be really sure though before loading it on your handset. It would be a shame to "brick" it.

Something to consider is all these phones use the G.729 codec, and unless their using the public domain implementation (which I doubt), there will be some licensed code for that in there somewhere which won't be in their released source code.

meric writes...

I'd be surprised if it is all there. Snom do something similar for their IP handsets, but the published code is really only the Linux OS. The actual phone application is proprietary code and not released.

For me it's hard to say what is proprietary and what is open source, and my friend only had a cursory look. I do know it at least uses the GNU oSIP library and iksemel (XML parser?) for the jabber client.

The first thing we want to do is to see if we can manually make the current firmware, i.e. no changes in the source code, then we'll re-examine it to see what can be changed.

It would be a shame to "brick" it.

yes, it is at the top of my concerns.

Z

Euro to Australian power adaptors

In my original post I indicated that I bought my hardware from Germany and consequently I had Euro power connectors on the AC adaptors. I indicated that I bought the cheapest converters possible on Ebay from seller ozfleamarket. The seller now only sell wholesale quantities (item 190205780370).

What can I say .... they are crap. Because they were so cheap luckily I didn't lose much, but for anybody else buying from Europe don't waste your time with these cheap ones. I spent a little more and bought these www.itsdirect.com.au/com...sub_category=146 from ITSDirect. It is not the only supplier out there but were the cheapest that actually carried stock. It is far superior than the Ebay junk consisting of a fully sealed moulded plug, with insulators on the on the power pins. It holds the Euro plug much more firmly and plugs into the Australian power socket much more securely. I should have bought these from the start.

Z

Reporting bugs to Siemens

This as it turned out was not an easy task. There is absolutely no where to do this, and consequently I was not hopeful that Siemens would get the bugs fixed.

The Gigaset website does have a contact from but when you select the product C470IP you only get the option of Product Info for Topic and Subject. This was clearly not the place as I have no need for product information so I tried to look elsewhere. I even tried Siemens locally here in Australia but no luck.

Out of desperation I decided to use the contact form to find a way to report bugs, and selected Germany as the country, as I did not want the contact form to be routed to somebody here in Australia and get lost in their black hole.

To my surprise they actually got back to me and said I can give them feedback through them via the ticket that was opened under Product Info. I reported and explained at length the MWI bug and also reported the Email address bug and the RSS feed issue as well. I got a response from them that they have forwarded my feedback to their development department and to check back occasionally for a new firmware.

Not much more I can do here, other than hope for a new firmware release soon that fixes the bugs.

Z

Hello,

I send you the answer of Siemens I hope it help you?

Voipling

RE: Dialling Plans for Siemens C470 IP
Just wanted to know what is the maximum number of dialling plans one can configure?

Dial Plans

I thought I'd post a section on dial plans so that I can answer jab08's question above as well.

This section is reached via the web browser interfcae Settings --> Telephony --> Dialling Plans

The dialplans on the C470IP are not particularly sophisticated like those that you can find in a Billion VOIP router or a Linksys 3102 ATA. They are really there for cost control only. There is no capability to modify the dialed digits in any sophisticated walk, like digit removal and insertion.

There are actually two sections being area code predialing and dialing plans.

Area code predialing is the only place where there is any digit manipulation. The C470IP allows you to set your local area code so that if you dial a number that does not start with 0 it will insert the areacode for you, which is necessary for most VSPs like Pennytel for example. However, it will not do so if you have a dialing plan, that is the area code predialing will never override a defined dial plan. My C470IP came from Germany and behaves slightly differently. It allows me to select the country and area code so that it can insert the international dial prefix, country code and area code. This was not described in the Australian manual so I assume it is not in the Australian version

The Dial Plan section allows you to control which of the telephone providers (VSP or PSTN) for a particular dialed number. Alternatively it can also block numbers. For example you could have a dial plan that says 0055 is blocked. The the dial plan simply works on best match. No capability to have wildcards, number of digits etc. For example if you have a dial plan with 02 and anther one with 029, and you tried to ring a Canberra number 026xxx, the 02 dial plan will take effect. I tried to test to see how many dial plans I can insert fro jag08 but could not find the limit (I actually inserted 21 dial plans before stopping).

For me I don't use this feature at all as all my dial plans are configured in my Asterisk server, however, the dial plans should be good enough for most, particularly if you have only one VSP.

Z

I just bought a C470 IP phone, I was after the C475 IP (with answering machine), but according to Siemens Australia, they'll be available sometime this year (soon ?), but no ETA or estimated price; so I got the 470 so I could get rid of the clutter of 2x phones and 1x ATA on my desk.

I have it setup ok with PSTN and my VoIP account at MyNetPhone - all seems to be working fine - receive PSTN fine, make VoIP fine - call clarity is good.

Now the tricky bit; I want to import most of my Outlook contacts into the handset's directory (most, because I have 297 and phone only takes 150 max in its diectory)

Outlook (2003) can only create a vCard one at a time, so I need a 3rd party application to export the lot, can anyone recommend one that will let me export all contacts to a file so I can edit out the ones I don't want, and then import them into the phone via the web configurator.

charonred writes...

Outlook (2003) can only create a vCard one at a time, so I need a 3rd party application to export the lot, can anyone recommend one that will let me export all contacts to a file so I can edit out the ones I don't want, and then import them into the phone via the web configurator.

Have at look at at one of my previous posts here /forum-replies.cfm?t=927308&p=2#r23.

The vCard import does not work very well so you are better off exporting your address book to a comma delimited file and build your own vcard file that the C470IP can handle properly. Just try to import that single vcard and see what it looks like on the phone.

I only had 50 or so to do so it was easier for me to cut and paste into a text editor than to build an automated process. If you wanted to build an automated process it is not that hard. just export an existing phone book and you can see the fields it requires.

Z

Yep zForum69, I read your post and have been searching for some 3rd party software to export Outlook to a file that I can work with; from what I've downloaded and 'trialed', I'm probably better off creating a text file from blank and filling in the details :-(

Unfortunately, I don't think the C470 IP quite lives up to its marketing hype. Whilst it's VoIP features are pretty impressive, without the answering machine of the C475 IP, it fails as a phone in some respects.

For one thing, the ring volume is pretty low even at max, so it's easy to miss calls if not in 'easy earshot'.

And then, it doesn't even show a 'missed call' notice on the screen; instead I have to check my messagebank service to see if I missed calls and if messages were left - not very efficient or convenient.

I've been through the pdf manual and web configurator, but there doesn't seem to be anyway of getting notification; a simple thing like 'missed call' - I made some missed calls and left messages from another phone, but nothing shows on screen, nor in the 'messages' section.

So I'm thinking 'back to hardly normal' and get a Panasonic KX-TG6700 ALB 2 line with answering machine which still needs the ATA for the one VoIP account (how many do I really need), but at least it'll let me know of missed calls and take messages; I can still make cheap VoIP calls, and I can then offload the $6/mth messagebank service.

charonred writes...

without the answering machine of the C475 IP, it fails as a phone in some respects.

That's why the C475IP is there. I just can't understand why it is taking so long to release it here locally. Same issue with the handsets. Because I didn't need it as the function is elsewhere I did not want to pay for something I will not use, thus the C470IP is ideal for me. If Australia ever gets the C475IP it should cost about 10% more than the C470IP based on prices overseas.

And then, it doesn't even show a 'missed call' notice on the screen; instead I have to check my messagebank service to see if I missed calls and if messages were left - not very efficient or convenient.

Something wrong there. I always get a missed called notification with the message lamp flashing away, so there is no way I can miss it as I described in the MWI post /forum-replies.cfm?t=927308#r5. I can't even think of what you might be doing wrong because it always worked for me without configuring anything. Are you on the latest firmware?

charonred writes...

So I'm thinking 'back to hardly normal' and get a Panasonic KX-TG6700 ALB 2 line with answering machine which still needs the ATA for the one VoIP account (how many do I really need), but at least it'll let me know of missed calls and take messages; I can still make cheap VoIP calls, and I can then offload the $6/mth messagebank service.

Or you could buy the unit you really need from overseas, the C475IP, and get some additional handsets while you are there.

Z

I think the 'missed call'/message will probably work with the VoIP line if I had a DID number, but I don't.

I simply have a MyNetPhone supersaver prepaid - cheap calls, but no phone number, and hence no VoIP voicemail message service - just not needed.

Incoming calls are on my PSTN landline with messagebank (not messages 101). It's a landline number I've had for years, it's on my business stationary, and will work with a bog-stock analogue 'telstra' phone if power is out.

Because the phone is setup with VoIP as outgoing default, I don't even get the 'broken' dial tone to tell me there's a message on the PSTN line - I have to dial 101 on PSTN to access messagebank, then enter my PIN before I know if there's a message.

If I swapped it around to PSTN as default, then I'd probably get the 'broken' dial tone and missed call/message, but then I'd have to prefix all outgoing calls to go out on VoIP instead of just dialling a number.

Maybe I could get C475 IP from overseas, which would solve the whole missed call message problem; but with delivery costs on top, and delivery time, plus power plug adaptors - and I only need 1 handset, so I think it's just too much fiddling around for what I need.

Don't get me wrong, I think the C470 IP is a great VoIP phone; but for anyone that may be considering a 2-line phone for PSTN & VoIP, then I'm just saying there are some things to consider re the standard landline and features they may have with their service provider, which aren't catered for very well by the C470 IP phone.

charonred writes...

I think the 'missed call'/message will probably work with the VoIP line if I had a DID number, but I don't.

I thought you hit the nail on the head there as that was a scenario I did not test. However, I just tested it by connecting the PSTN line straight to the C470IP and it worked. I thought it was because I am paying for caller id on the PSTN line, but I dialled again this time blocking my CID on from my mobile phone, and it comes up as Withheld but it still turned on the message lamp. I'm happy to help you diagnose your problem by trying something on mine if you want.

charonred writes...

Incoming calls are on my PSTN landline with messagebank (not messages 101). It's a landline number I've had for years, it's on my business stationary, and will work with a bog-stock analogue 'telstra' phone if power is out.

I totally understand. I'm reluctant to release my PSTN line until I have some sort of confidence with LNP, and I don't even use that number for a business.

If I swapped it around to PSTN as default, then I'd probably get the 'broken' dial tone and missed call/message, but then I'd have to prefix all outgoing calls to go out on VoIP instead of just dialling a number.

I don't think that will help. Since it is an IP phone it works much like a mobile phone. i.e. you dial then send, you don't wait for a dial tone before sending.

Maybe I could get C475 IP from overseas, which would solve the whole missed call message problem; but with delivery costs on top, and delivery time, plus power plug adaptors - and I only need 1 handset, so I think it's just too much fiddling around for what I need.

It did not turn out to be that painful. Cost including delivery is still 10% cheaper than the best price locally. The power adaptor turned out to be quite seamless, just don't get the super cheap ebay ones I originally tried but the other ones that I replaced them with. Delivery was a bit long. No warranty is also something you need to consider.

If you are considering the C475IP you must consider the language of the voice prompts. Somebody made a valid point that they may not be in english. I bought mine in Germany so this would be a concern for me if I was looking for a C475IP and needs to be checked. Otherwise you could always look for a cheap online store in the UK.

charonred writes...

Don't get me wrong, I think the C470 IP is a great VoIP phone; but for anyone that may be considering a 2-line phone for PSTN & VoIP, then I'm just saying there are some things to consider re the standard landline and features they may have with their service provider, which aren't catered for very well by the C470 IP phone.

Thanks for your input. You made some very valid points that others who are considering buying this phone need to consider.

Z

I just wish the price would drop a little bit more :-)

spunkarooney writes...

I just wish the price would drop a little bit more :-)
Check this out. Siemens
Gigaset C475 IP DECT VoIP Telephone
£74.99 equates to $159 at today's rates.
www.cordless-phones.uk.c...arch_text=C475ip
Can't understand why we have to pay $299 for the same here!!!!!!!!!!!
www.telephonesonline.com...hp?page=1&q=C475

I just got the C475IP from Hip-IP and it arrived in five days which was fantastic, i see they have the S685IP

www.hip-ip.com/product/v...85-ip-dect-phone

in stock now, i am tempted to get this item too its so cheap! Why can these phones not be as readily available in Australia!

lwaites writes...

I just got the C475IP from Hip-IP and it arrived in five days which was fantastic

Very good price, particularly with the excellent exchange rate we have at the moment. Definitely the way to go if you want the in built answering machine just in case there is a language issue. Which package did you buy and what was the total cost (AUD$) to get it delivered here in Australia?

i am tempted to get this item too its so cheap!

I'm still skeptical about High Definition Sound Performance on a telephone but the bluetooth feature is nice. Come to think of it, if the S685 was available at the time I think I would have bought it as well.

BTW what did you use to convert that big clunky UK plug into an Australian plug?

Z

I just got a simple converter plug, think it was from ITSDirect. Its the bluetooth that sells it to me, to have a phone with voip capabilities, bluetooth and an integrated answer machine is a first as far as i am aware.

Hi there from the UK.

I came across this thread a couple of months ago when I was searching for a similar phone. Thanks for the detail and all the other posters. It helped to clarify my decision no-end.

I have recently purchased a triple set of the S685IP and have done a review with screenshots/photos etc on my blog here: www.theopensourcerer.com...set-685ip-phones

My impression so far is very good indeed.

Thanks again.

Alan

Hello from California,

Will this S685IP be sold in the US anytime soon? If I were to purchase from over seas as others above, will it be compatible with the VoIP service offered here in US?

I was sold on the specs of this system for my small home base business.
What would be a recommended alternate if this Siemens S685I?

I've read so much these past few days......

Thank you,
jaf

Jaf,

The phone should be compatible with any VoIP Service Provider that uses the SIP protocol. (covers most except Skype)

However I believe that the DECT band in the USA is 1.9 GHz while the Europe DECT band is 1.8 Ghz. It should still work providing you get any additional handsets from European sources. You may cause interference to or get interefence from whatever equipment uses the 1.8 Ghz band in the USA.

Another issue may be the plug packs. I have an S450IP and it uses 240 volt plug packs, not universal 110-240 volt plug packs.

Trevor

TrevorB writes...

However I believe that the DECT band in the USA is 1.9 GHz while the Europe DECT band is 1.8 Ghz.

I think you are spot on there. I think this is the reason there is a delay with any introduction into the US as different electronics suitable for the 1.9Ghz band are required. They are labeled as DECT 6.0 phones.

Another issue may be the plug packs. I have an S450IP and it uses 240 volt plug packs, not universal 110-240 volt plug packs.

Same here. The ones I bought from Germany are not multi-range plug packs.

Z

Hi Jaf,

Production for the US model has not started. As far as we are aware Siemens are trying to gage demand prior to sanctioning production. We get quite a few enquiries weekly about shipping the model to the States.

If I hear of any developments I will let you know, but I am not sure the model will be available anytime soon.

JP

Thanks to all for the info. Sadly, I'm not up to the challenge of taking the chance, or patiently waiting and hoping it'll arrive sooner.

A lesser alternative here seems to be the Siemens S450 IP.

???

Jaf in Calif.

jaf writes...

A lesser alternative here seems to be the Siemens S450 IP.

I think you will be quite satisfied with that choice.

Z

The S450IP is still a solid model - I used it up until the launch of C475IP and was more than happy with it.

JP

Hi all,

I would like to add to this post praising the C470IP.

I have recently changed from using a SPA3102 with a Uniden 8855 cordless handset, and the C470IP is far superior.

After years of testing and refining I could never get the SPA3102/Uniden combo working together perfectly. I always had some echo, and I could never get 3 way chats working adequately.

The C470IP has fixed all those problems and has given me a number of handy new features to boot. The biggest change is the improvement in call quality. The call quality has improved significantly.

I tried to buy from Ryda and Prestige Communications but both vendors told be that it was indefinately backordered. In the end I bought them from FreshTel for $199 ea. FreshTel was very good to deal with even though I told them I did not intend to be a voip customer.

I hope that indefinately backordered does not mean that this product is slowly being discontinued as it is really very impressive.

The phone came setup for FreshTel as a default but it was very easy to change settings to my existing provider (Maxo).

In all I would recommend the C470IP to anyone who is looking for a cordless voip phone.

Cheers

Hi zforum69 et all,

I would like to know if the phone has call forwarding features?

For example:

There is an incomming call on SIP #1. (DID overseas)
1. Ring on the handset (home)
2. If handset not answered then forward the call to mobile via SIP #2 (local voip provider).
3. (as a bonus) If mobile is not answered then send to voicemail.

1 and 2 are crucial as SIP #1 blocks all online routing applications such as voxalot and 90% of the time I'm away from home.

Thanks.

dmainou writes...

There is an incomming call on SIP #1. (DID overseas)
1. Ring on the handset (home)
2. If handset not answered then forward the call to mobile via SIP #2 (local voip provider).

I think so, but I have to find some time to test it out for you.

Z

Hi,

I have 6 of the S685IP phones - here in the UK

I am trying to forward all incoming landline calls, when they are not answered to another number using one of my VOIP accounts but I cannot see a way to do it. The only accounts I seem to be able to forward are the VOIP accounts - not the landline account.

Am I missing a trick?

Thanks

Jon

dmainou writes...

There is an incomming call on SIP #1. (DID overseas)
1. Ring on the handset (home)
2. If handset not answered then forward the call to mobile via SIP #2 (local voip provider).

After much testing my guess that it would work they way you described was incorrect.

It seems the call forward feature is a function with the SIP provider that the call came in on. That is when the call comes in on SIP #1, and it forwards after no answer it sends that request back SIP#1. That is, the C470IP DOES NOT bridge the call and therefore will never make the call via SIP #2 or any other provider. The other issue is that your VSP must be able to support this, as the C470IP is actually not doing the forwarding, your VSP is doing it. After much testing I found iinet and Pennytel do not support this.

I was able to confirm this is exactly how the C470IP does it by connecting to my trixbox and seeing it all happen. Unlike my VSPs my trixbox does support a SIP call redirection request.

It took me a while to work out what it was doing as I expected it to bridge the call. This was certainly an eye opener for me on how it works.

Z

jonsinclair writes...

Am I missing a trick?

No, you are correct. The call forward function on the C470IP are SIP redirection requests and therefore not relevant to the analogue line (see previous post).

Z

Many thanks for your reply and for all the tests that you have done.

Jon

Anyone got any experiences with the snom m3? I'm tossing up whether to go with the siemens or the snom.

anyone know if you can transfer calls from one handset to the other? if so how?

btw, cheapest UK for the C475 I found was http://www.broadbandbuyer.co.uk/Shop/ShopDetail.asp?ProductID=6617

just thought you'ed like to know as you've linked another site above thats more expensive.

Hi All,

The C470 is on sale right now at Myer for $199. No extra handsets though. I did learn from looking at the box that there are a number of Siemens Gigaset handsets that will work with the unit – not just the C47H.

Cheers, Cindy

CAP-E65 writes...

The C470 is on sale right now at Myer for $199

Is that branded with a VSP like Freshtel? There were earlier references that suggested the first 500 were not locked but any future ones will be

I did learn from looking at the box that there are a number of Siemens Gigaset handsets that will work with the unit – not just the C47H.

It will work with any DECT GAP compatible handset, but you lose all the IP features on the handset, e.g display weather, transfer address book, email notification etc, as they have have nothing to do with DECT, other than Siemens implementing a proprietary way to use the DECT channels as a data channel.

Z

jag08 writes...

My comment was in relation to comparison of calls between IITALK & PENNYTEL using the same codec G.729 .

G.729 is good quality on Pennytel compared to G.729 on IITALK. To get the resonable sound quality on IITALK I am having to use G711 ulaw. I don't know the reason why calls via Pennytel sound better even on lower codecs. Perhaps someone here can explain that.

I'm interested to know if your able to use the one iitalk/iivoip account between 2 or more handsets and are able to make multiple outgoing and incoming calls via the one account with different handsets, without the need for a trixbox (as i understand with iivoip u can only have one registration for your account)

Also are you able to have dial plans to use pennytel just for mobile phone calls and over seas

VeNT writes...

anyone know if you can transfer calls from one handset to the other? if so how?

I assume same as s685IP, as most things the same. Simple. Nav key right click, scrollup/down to select which handset.
Manual is downloadable from gigaset.siemens.com

Mike 802 writes...

... are able to make multiple outgoing and incoming calls via the one account with different handsets

You can certainly do this with Pennytel if you subscribe to this option. Requires auto top up. We have conferenced two lines to different corners of the globe, no problems (on s685IP).

Also are you able to have dial plans to use pennytel just for mobile phone calls and over seas

Yes. Dial plan works simplistically from lists of prefixes or complete phone numbers eg 04 direct one place and 039 another. 0011xx another, etc. As many as you like. You can nominate any default provider for the rest.

Mike 802 writes...

I'm interested to know if your able to use the one iitalk/iivoip account between 2 or more handsets and are able to make multiple outgoing and incoming calls via the one account with different handsets, without the need for a trixbox (as i understand with iivoip u can only have one registration for your account)

The short answer is not really. There reason being is that iinet try to make their product look like a PSTN line. As soon as there is an active call via iinet, any inbound calls will see the line as "engaged" and sent to voice voicemail.

You can however you can make simultaneous outbound calls. Don't try to explain this to the helpdesk though as they are obviously trained to make their VOIP service look like a traditional telephone. Only after I highlighted to them on their call detail records, two calls made at approximately the same time and their duration indicating a simultaneous call, did they concede.

Basically if want to use VOIP beyond making it look and behave like a PSTN line, don't use iinet.

Z

Just press the 'int' button on the left hand side of the square in the middle of the phone, then select the handset you require.

JP

PS – we do offer price match on all Siemens Gigaset IP Phones.

JP,

Your prices for the C475IP are OK but if you can reduce your postage rate from 39.99 pounds to something more reasonable you may get more sales from Australia.

I have bought similar size devices from the UK and the postage is about 20 pounds for Royal Mail air express.

Trevor

anyone know if its poss to link two bases? I've a network spanning two offices (both with about 6 people in) at different ends of a building
we've a ethernet connection between the offices and a gigaset in the main office
there is almost enough signal for the gigaset to work in the other office but it keeps dropping (other people use the building with wireless etc so I'm guessing its just interferance) but I was wondering if I bought another base station would I be able to use them togeather with 6 phones, ie be able to make intercom calls etc between the offices?

You may need to look at something a bit less 'resi' with repeaters. Something like the Kirk DECT (I think Polycom just bought them a while back...).

I think the gigaset only allows 2 concurrent calls (although you can have 6 handsets) — or is that 3 with the pstn port... hmm...

I think Astra also have some SIP/DECT office grade products, but the Kirk stuff has been around a loonggg time...

the number of calls isn't an issue tbh, 2+1 (VoIP/Landline) its more just having enough signal in both offices.
I'm gonna move the box around a bit and have a go to see if I can get it any better.

zforum69 writes...

Is that branded with a VSP like Freshtel?

No the $199 price at Myer for the C470 is not locked to a VSP. Not sure when the sale ends but I assume it is a end of financial year sale.

TrevorB writes...

Your prices for the C475IP are OK but if you can reduce your postage rate from 39.99 pounds to something more reasonable you may get more sales from Australia.

I ordered a Siemens s685IP quad (base & 3 extra handsets) from hip-ip, and the postage cost was just over 19 pounds (Fedex to Australia), so perhaps they have listened to you. Speed was excellent too – ordered on a Sunday, arrived the next Friday.

Hi All,

Just adding my input to this topic...

I bought 2 C470IP units a few months back, and I am very happy.

I have been using SPA3000 for a few years, and that was a good unit. Much more configurable than the C470IP, however at the end of the day ti is an analog telephone adapter, and I had to do much tweaking with gain settings and power levels to get the sound on the telephone handset to what I regarded as close to straight PSTN quality. (What a good learning experience it was configuring an SPA3000!)

The c470IP removes this hassle, as it does not have to interface the mic and speaker in and out of a 2-wire analog connection between the handset and the voip unit, and as a result the voice quality is good to start with no effort required, and stays that way. Too easy!

Additionally, the unit supports – and registers – up to 6 voip accounts simultaneously. I use 3.

I have done conference calls between two external voip connections, one overseas, and it works fine. (but no mute key – why not !!)

Handsfree works well too.

The nicest surprise feature was that the address book can be downloaded, uploaded and deleted from the web interface to the base station – independently for each handset. I have even used this on an older Siemens DECT GAP handset (can't recall model) which was supplied with a bog stock PSTN base, which was able to register to the 470IP base and work fine.

My main complaint is that the dial plan is a bit limited, no substitutions etc. Sipur is much more capable. I have been able to do some workarounds but this is less than ideal. Probably not a problem if you have only one voip account, and may not be a problem in all cases anyway.

Also, the two hard keys under the display on the handset can be configured to do different things. To make things simple for the family I have configured the left hand key to "IP" and the right hand one to "Fixed Line" and they can simply press either before dialling and this overrides the dialplan.

One last thing I could not get working was the message waiting – but I have not really tried this out much. I tend not to use voicemail.

Hope this is useful.

D

Oh, and another thing that I have discovered about the C470IP – it does not seem possible to set up a voip account as a gateway account for outgoing calls without the unit registering that account for incoming calls as well.

Sorry fellas,
Bit off topic, went out and got one of these phones today! (c470ip) all seemed good except that no matter what phone i ring, be it another iinet voip phone or pstn i get cut off at the 40second mark and get a 701 code (other end cut off call) anyone had this. Im with iinet naked.

After talking with iinet they assured me it wasnt them and now i cant even register. Havnt changed any settings.

im already over it and have had it for a day!
any ideas??

EDIT – ok now its just saying 'terminated' and calls are only lasting 36 seconds.....weird that every call lasts exactly the same.
Gunna check router/phone for anything that resembles 36.

zforum69 writes...

You can however you can make simultaneous outbound calls. Don't try to explain this to the helpdesk though as they are obviously trained to make their VOIP service look like a traditional telephone. Only after I highlighted to them on their call detail records, two calls made at approximately the same time and their duration indicating a simultaneous call, did they concede.

I just got myself a twin pack of the S685IP and setup my one IInet account to be used on both handsets as the default account when trying to make 2 calls at the same time from different handsets I get a VoIP error code 400 on the 2nd handset that tried to make the call after the first one

Is there any way for this to work as u highlighted above or is there may be an auto fall back option for the 2nd phone to use my pennytel account when the iinet account is in use

at the moment i'm able to make 2 out going calls via my one pennytel account using both handsets not sure way iinet isn’t liking my current settings :(

Solid Rock writes...

Bit off topic, went out and got one of these phones today! (c470ip) all seemed good except that no matter what phone i ring, be it another iinet voip phone or pstn i get cut off at the 40second mark and get a 701 code (other end cut off call) anyone had this. Im with iinet naked.

It sort of sounds like a funny NAT/ALG type issue probably caused by your helpful router/modem ;)

http://icm-km5.erlm.siemens.de/docs/cp/attachments/VoIP/nat-troubleshooting_en.pdf

I found that setting the c470's 'bind' port (ie the port it listens on) from 5060 to 49060 fixed this up, no crappy port forwards etc (my Billion was doing nasty things to the SIP packets).

YMMV.

ok so i the sip/rtp ports to what you suggested in the advanced tab.
do i also change:

Proxy server port: 5060
Registrar server port:5060
Outbound proxy port:5060

do i change them to 49060?

if i do the phone wont register, even if i port forward them.

hmmm annoying.

EDIT – FIXED!! haha After an hour of looking through the posts on the linksys website i found a bloke with a similar problem. (Calls only lasting 30 seconds) What i did learn is that firmware 1.00.11 for linksys wag325n is buggered, most people use .8 or the new .12.
So i upgraded to .12, put the 5060 ports in the "port triggering" and enabled qos. And yea calls now last more then 30seconds.

Thanks all for your help

Mike 802 writes...

Is there any way for this to work as u highlighted above or is there may be an auto fall back option for the 2nd phone to use my pennytel account when the iinet account is in use

When I tried this out it was not with the Siemens, from memory I think I did it with a couple of PCs with softphones.

There is a possibility that iinet removed even this capability to REALLY make it act like a PSTN. I don't know when I'll have the time to try it out, but if you are willing to give it a go, try making simultaneous another way: eg 2 softphones or a softphone and siemens. If that works then try to work out how to do the same with the siemens.

Z

I love Whirlpool, always get the info you want to hear! I can't believe this type of product exists. Now I can get rid of my 2x SPA3102 ATA's and 4x Panasonic Cordless phones.

###### JB HIFI has the phone on discount ######

$169.99 as an end of financial year thing.

I bought it last nigh and I'm impressed with the quality of the phone.

dmainou writes...

###### JB HIFI has the phone on discount ######

$169.99 as an end of financial year thing.

Which store?

The one i went to wasn't and was still @ 199.

They are in the latest JB Catalogue as $199 bundled with a FreshTel USB Stickphone.

I got one today at Chadstone (Vic) store for $165 after telling the sales assistant that I believed the were on special for $169 as per post above. Anyway he said he could do it for $165 (after consulting with his PDA) without the USB Stickphone. Since I already have a Stickphone it was not an issue.

Upgrades the firmware to the latest and the configuration look identical to my older S450IP.

Trevor

Just wanted to thank you zforum 69 for your detailed insight into the C470 IP – this is by far the most helpful info out there on the www.... Also, thanks to Am going to buy from JB Hi fi today is possible....

Just one question I'm wondering if you could please help with?: is there any way of turning the C470 IP with additional handsets into a 2 fixed line system (I have two incoming phone lines but only use one to dial out- the other is for faxes out and incoming calls only)

Any advice here would be much appreciated – like many on this forum I have tried to contact Seimens with no success (apparently only one gentleman deals with Gigaset calls for the whole of Australia!)

Just to let you know – JB Hifi are unable to take phone orders and post out products, so if you don't live near a store (I'm in Cairns) the next best option seems to be Myer Brisbane at $199 plus $8.80 postage....
Freshtel offer the C470 IP for $199 plus $19.95 but you have to sign up for 12 months on a $9.95 Freshtel plan which doesn't suit me...

metropolis writes...

(but no mute key – why not !!)

I am now the happy owner of the Gigaset S685IP phones with 4 handsets thanks to the great research by z and the quick delivery from Hip-IP in the UK (8 days from order to delivery). All the great qualities I read about here are true.

The only major flaw for me is the lack of a mute function. The owners manual says the feature is there but it works in this way – you press the control key on the left (marked INT) and it places the external party on hold not mute. Sure the external party can't hear you but you can't hear them either all you hear is loud obnoxious beeping until you hit the return key to get the call back. The manual says that the other party hears hold music but you are able to turn off that "feature". This is a major show stopper for me since I spend hours on teleconferences during the week and I don't want the other participants hearing everything going on in my house.

I sent a note to Siemens to make sure that I was using the feature correctly. Here was their reply: "In regards to the query sent about GIGASETS685IP, we would like to inform that is normal function of the phone, and is not changeable that has been designed the way it is. "

Other than that I am still getting to know the phone better but happy so far.

Cindy

notsureaboutit writes...

Just one question I'm wondering if you could please help with?: is there any way of turning the C470 IP with additional handsets into a 2 fixed line system (I have two incoming phone lines but only use one to dial out- the other is for faxes out and incoming calls only)

I personally cannot see how that could work ... sorry.

Z

CAP-E65 writes...

The only major flaw for me is the lack of a mute function.

Since you said you use the phone for a lot of teleconferencing, I would expect that you use a headset as well. I think your only solution is to get a headset that has a mute switch.

Since you have a S685IP, you might even use a BT headset so you can go off and make that cup of coffee with out carrying the handset and other cables. Again make sure that BT headset has a mute function.

If you do happen to use a BT headset, let us know how it goes, I'm sure others beside myself would like to know.

Z

Freshtel offer the C470 IP for $199 plus $19.95 but you have to sign up for 12 months on a $9.95 Freshtel plan which doesn't suit me...

Hi notsureaboutit,

It's currently $149 + P&P for a C470IP from Freshtel if you sign up on one of the new Freshtel plans (which start from $4.95).

Regards ,

– David.
Freshtel Pty Ltd.

metropolis writes...

Oh, and another thing that I have discovered about the C470IP – it does not seem possible to set up a voip account as a gateway account for outgoing calls without the unit registering that account for incoming calls as well.

I have since emailed Siemens support, and they have confirmed that it is not possible to set up this unit to make outbound voip calls without registering first with the voice provider. This I could do on the Sipura 3000/3102.

If anyone figures out a work-around to fool the unit into achieving this, then I would be very interested to know how it is done.

D

I got my C470IP last week, and after waiting for Siemens to post me the power supply (not included in the box??) I am up and running. Thanks for all the info, it was a breeze to register it on my Asterisk box and start making calls. Good quality and range, so far no problems!

Ok i don't have this one nor do i use it. However i think the Handset does not support G.722 though the C470 IP itself can, reading through the Manual. If you do purchase this thinking you can have G.722 i'd say yes but provided you one of the handsets supporting G.722. The S685 IP can.

Please note there aren't many providers supporting G.722 yet, apart from the Siemens gigaset network, nor would you be able to have a conversation in G.722 with PSTN users.

Also saw that JB Hi-Fi have now officially display it as A$169.00

I wonder if this unit is good because it doesn't have to do Transcoding of the codec twice which i think happens when you have a ATA and a Handset(Uniden etc). The handset is always using G.726 and so if it gets converted to G.729 or G.711 or G.726 i think it gets decoded and coded again. If this unit is used with a different handset say like Uniden i think transcoding is likely to happen. Does anyone know?

I'd be interested if anyone who has experienced better quality than using the ATA kindly advise in terms of if the voice is more rich etc. At the current price it's looks good to buy though hopefully additional handsets become available.

Thanks.

Cheers

P.S: Some info for those wanting to save power... Bought one of those Energy Meters @Jaycar and they are an eye opener. Shutdown PC uses about 30 watts. Now i should buy one of those Powerboards which you can switch off remotely so no power is available to the slots you want to be switched off for good.

Hi All, I've just heard from Siemens re the expected release date for the additional C47H handsets for the C470 IP: due to be released in Australia mid August 08, and will retail for approx $90....

Will be eagerly awaiting this date....:)

zforum69 writes...

If you do happen to use a BT headset, let us know how it goes, I'm sure others beside myself would like to know.

Z, Yes I have now tried the Bluetooth headset feature on my S68H handset (S685 IP base). It works just fine. Once a call is dialed you can press the appropriate key on the headset and it transfers the call from the handset. The range was good as well as I could go into the kitchen to make a cup of tea while the S68H handset was still in my study. To get the call back to your handset you just need to press the speaker phone key or call key and it takes it right back.

There is also a feature of the S68H to sync your data with your PDA via Bluetooth but I can't imagine why you would do that since the Web Interface also has the ability to transfer your phone numbers to the phone from a file on your PC. Seems like the web interface is a more direct way to go.

On the topic of acceptable vCard formats, I have found that the vCard 2.1 export from my Lotus Notes address book transfers my phone numbers without a hitch once I delete the note field from the file created. I am not sure if all notes create a problem or only the long ones which I seem to have a lot of.

Cindy

Hi,

FYI, the Siemens phone Freshtel are selling at JB HiFi and on line are not locked to Freshtel.

Simon

CAP-E65 writes...

There is also a feature of the S68H to sync your data with your PDA via Bluetooth but I can't imagine why you would do that since the Web Interface also has the ability to transfer your phone numbers to the phone from a file on your PC. Seems like the web interface is a more direct way to go.

Gigaset Quicksync (as far as I can work out) only works via Bluetooth connection to a S68H handset (s685-IP). The software is downloadable from Gigaset site & not included in the box or supplied CD.

There is also a "Send To Bluetooth S68H" on the menu. The standard Outlook export did not look compliant, so the sync was easy.

I just purchased a couple of IP Phones, a C470IP wireless and a Linksys SPA942 desk phone – the C470IP for general domestic use and the SPA942 for SOHO use. Was pleased to discover my existing Uniden HS-910 headset works fine on both units.

First thing I have to say the SPA942 is absoloutly brilliant, I really can't pick any faults with it at all. I was a little disappointed with the C470IP – viz:

1. The microphone in the C47H handset could be better quality – if you plug in the Uniden headset its fine, but the in-built microphone is a bit fuzzy and very sensitive to breathing noises.

2. There is no way to adjust microphone gain.

3. No "mute" facility (this has been mentioned previously).

Also, when setting up the C470IP base station, you may have to move it around to eliminate RF interference from other computer equipment, not a big deal, but worth mentioning. In summary, I'll be keeping the C470IP, but I won't be rushing out to buy additional handsets as I think this is the unit's weak point.

BTW, I purchased my C470IP (Freshtel) from JB HiFi at Miranda for $180 (marked price $199).

Just purchased a C470 today from JB HiFi. Unfortunately the Base Station power adaptor was missing from the box, a situation that a previous poster also experienced.

A quick call to JB HiFi revealed that several of the C470's were missing Base Station power adaptors. Finally an adaptor was located, after opening several boxes.

So buyer beware, before leaving the store with your C470, open it up and confirm you have a Base Station power adaptor, as it seems a lot of units are being supplied without them.

Starwraith writes...

eems a lot of units are being supplied without them.

Yes we too were supplied one without base station power supply last week.

We have had a few customers with exactly the same problem, must be a problem on the packing line.

Bricked C470IP

Just a quick note on how I "bricked" and then managed to "unbrick" my C470IP. Started the built-in firmware update procedure using the web interface, went away from my desk and came back 15 mins later to find the base station unresponsive to the web and the phone handset unable to connect to the base station. Power cycled everything, tried full resets, everything dead (bricked although basestation still pingable on the net).

In the end it turns out that when you click on the firmware upgrade, the phone wipes it's firmware leaving only a very basic firmware upgrade system left on the device. All it does is attempt to contact the gigaset website to download new firmware, if it can't get to the website, your phone will just be dead until it can.

In my case, the problem was that the TCP/IP stack on the firmware loader is very primitive and it was being blocked by my firewall (pf on freebsd) because it was seeing the packets as "bad". Ended up moving the device to just a bog standard adsl modem and it downloaded it's new firmware in about 3 mins and was up and running again.

Hope this helps someone.

Tony

Where can I get the C470IP + 2x additional handets for cheap?

You can get the C470IP from JB hifi for $165. (well we did anyhow last week).

Additional handsets cheap, I am not sure I am looking for 1 additional handset too.

metropolis writes...

I have even used this on an older Siemens DECT GAP handset (can't recall model) which was supplied with a bog stock PSTN base, which was able to register to the 470IP base and work fine.

With a non-voip DECT GAP handset, how do you select the outgoing line (pstn or sip account) before you dial?

eug k

- suffix the called number with #0 for PSTN or #1-6 for any of the 6 available SIP accounts to select the outgoing line.

You can set the default outgoing line to either PSTN or VoIP (globally – not per handset)

Trevor

In respone to eug k...

Exactly as TervorB says, suffix lets you select lines.

If your outgoing is voip by default, you can assign a different default outgoing voip line for each handset.

Also, you can also block/enable access to each individual line for each handset; can be set different for incoming and outgoing lines on each handset too.

Awesome, thanks! I might replace my SPA3k + various DECT handsets with one of those!

Stability-wise, are the Siemens VOIP dects rock-solid like the SPA series? I never have to reboot my SPA, it just works and works.

very solid from my several months of experience. Automatically re-registers if your router or line goes down – never had to power cycle it so far.

One catch if you have multiple SPAs and multiple voip accounts – the dial plans are not very sophisticated, and can't do things like substitute digits.

Whether this is a problem depends on how tricky you want to get. I have three voip accounts, and it isn't a problem for me. Might want to download the manual from the siemens web site and check it out if you think you might need to do something particular.

Also, where SPAs can use a number of voip accounts as gateways (place calls without registering for for incoming calls) the siemens cannot. I checked with siemens, and any account that is set up on the unit must register with the service provider. This might be an issue of you have multiple phones that need to dial out on one common account, but are too far apart to be on the same DECT base station. Bit hard to be sure where any incoming calls will go in this scenario.
Not a problem for me, but could be for some.

This phone has been great for me; I have not any problems at all. The hands-free from the handset is amongst the best I have used – including office PABX handsets – and we use it regularly. Just wish it had a mute button !!

good luck

anyone using these with a DG834 V2? I've heard the router has some kinda built in QoS for VoIP but I still get bad noise when using the internet at the same time.

Just got this phone from JB $142. Using Pennytel. And I do not have a fixed line all calls are VoIP calls. Wanting to know how to set a dial plan for mobile, 13xx, 1800xx numbers to send all these calls via Voip as this is the only service available.

Edit: Also how can you use the Penntel feautre SMS with this phone or does the SMS only work with fixed line.

I have one of these and 2 additional handsets. Have it setup to use Telstra as my fixed line (101 doesn't show you have any messages on the handset so you have to dial in to check. Does anyone have this working so that the handset alerts you to new Telstra voicemail.)

I also have two VOIP accounts to call back to the UK. One is Tesco (no caller ID but then again not sure if Tesco supports this. As with Telstra, Tesco has no voicemail alert on the handset. Does anyone have this working?

My other VOIP account is with Sipgate. This works great both caller Id and voicemail alert work as expected.

I've had these over a week now and no problems whatsoever. I installed a new firmware this morning (02123) that is meant to have an update to voicemail. Not sure what the update is meant to do but if I notice anything different I will post it here.

J

I also meant to add a thank you to zforum69. I have to admit reading your post inspired my purchase.

jamesjfa writes...

I have one of these and 2 additional handsets

Did you get this in Oz or from O/S? I want one of these, but only when I can get a second handset here.

FredBerk writes...

Did you get this in Oz or from O/S? I want one of these, but only when I can get a second handset here.

I bought the Main unit from JB-HI ($169) and then the 2 other handsets from the UK. Siemens confirmed that the handsets are not available in OZ yet.

It seems to be quite slow at resolving the names from the contact list when someone calls. More so with mobiles.

J

FredBerk writes...

Did you get this in Oz or from O/S? I want one of these, but only when I can get a second handset here.

That's only a price issue, as I am sure you know you could just buy another complete unit from JBHifi for $142 someone posted the other day ?. (we paid $169) and register the new handset as an additional to the original base station.

Bit of a waste of the 2nd base station, but it would solve your handset problem.

boofh writes...

That's only a price issue

Good point – hadn't really thought of it that way...

Bit of a waste of the 2nd base station

Just one problem – my home office/study is starting to look pretty geeky sad, computers, routers, ATAs, etc.. :-(

Does anyone know, if I bought two complete units and used the spare base station as a repeater to get better range, would I need to have a network cable to the second base station? It would be, say, 100m away at the other end of a factory.

Hi All, i'm currently using an netcomm nb9w for my internet and voip access, could i use this siemens unit with my nb9w as a plain network router and bypass the voip settings in my nb9w as the siemens unit sounds more reliable, currently awaiting Ndsl to be turned on with Exetel and then the savings really start flowing. your help and advise would be greatly appreciated. oops forgot currenly with MNF and very happy with them

Doff

doff23 writes...

Hi All, i'm currently using an netcomm nb9w for my internet and voip access, could i use this siemens unit with my nb9w as a plain network router and bypass the voip settings in my nb9w as the siemens unit sounds more reliable, currently awaiting Ndsl to be turned on with Exetel and then the savings really start flowing. your help and advise would be greatly appreciated. oops forgot currenly with MNF and very happy with them

Doff

I would have thought yes but I don' t know the specific method. I think you'll get more luck if you ask that question in a separate thread e.g. "How do you turn off VOIP in Netcomm NB9W?" as it really has nothing to do with the Siemens.

Z

Thanks to all the helpful information here, I went out and bought a unit last Sunday from JB for $142.

Interestingly, they had two "plain" C470 sets and three Freshtel branded ones. I brought one of the former home and was really surprised to find the mains charger missing. When I spoke to the folks at JB, they checked the other "plain" set and it also lacked a mains charger. In the end they gave me a Freshtel branded set.

Setup was really easy and it's working great with Pennytel.

Cheers

Yep just bit the bullet aswell and went to JB bought one for $142.00 it took me about 40Min to configure with MNF but now the connection is rock solid and much better call quality than i had with the voip on my NB9W this unit has not fallen over yet unlike my Netcomm every second call having to reset the damn thing otherwise it would redirect the calls direct to voice mail. now i have a happy wife ;-)

Just taken the plunge on mine. $139 from JB, showed the sales guy this thread and the $199 sticker price plummeted by $60. (IMO: JB cant be making money on that sort of deal).

Couple of things I found:

For some reason, the base station wouldn't connect properly to the ethernet port on my Cisco 837 router – the link status was good, but only about 1 in 10 pings would reply. And forget even trying to open the admin web pages.

QoS – out of the box, the C470 marks the outgoing RTP packet stream with the Differentiated Services 'Expedited Forwarding' (EF) code point, which means that you only have to prioritise DSCP EF packets at your router – no fiddling around with matching IP addresses or ports.

New Firmware it's available:

Gigaset C470 IP / C475 IP / S675 IP / S685 IP Firmware update 07/2008

Download version: 02123

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

Source: http://gigaset.siemens.com/shc/0,1935,hq_en_0_145510_rArNrNrNrN_variation%253A-5_pageType%253Adownloads_imagePos%253A0,00.html#content

Gigaset Sales Partner: http://shop.voipdistri.com/index.php?cat=c78_Siemens.html

I need a voip phone with bluetooth... will I lose any functionality by using an S68H handset with a C470IP base, as opposed to buying an S685 IP set? Are all the bases the same?

I don't need the answering machine.

The reason is that I don't wanna shell out full price for the S685 IP in case it doesn't work with my router... so if the JB-sourced C470IP works, i'll get an S68H handset from overseas.

No matter what I did, I couldn't get my SPA3000 to work with my ClarkConnect router – it wouldn't register. I switched to a Linksys WRT54G and it works fine. Now I've replaced that with a Cisco 857W and it refuses to register again. My Grandstream GXP-2000 has no problems connecting, and neither does my Nokia's built-in voip client over wifi. Grr.

thanks

My thanks too to zforum – your detailed posts made my search for a voip solution much easier (I ended up with a S685).

The only minor annoyance I have is with the irritating electronic music on hold – is there any way to change that, preferably for an MP3?

kko writes...

My thanks too to zforum – your detailed posts made my search for a voip solution much easier (I ended up with a S685).

may i ask where you got the S685 from and how much it cost?

thanks

I got my S685 IP (base station and four handsets) from hip-ip.com, a UK company, for 233 pounds and 19 pounds postage. Ordered it on a Sunday, arrived the next Thursday.

kko writes...

got my S685 IP (base station and four handsets) from hip-ip.com

Hi,

SO what is the difference, if you know, between the 475ip & the 685? Apart from 25 pound.

Does anyone know a way to reboot these without physically unplugging the base.

Mine was registered, and I could call out, but it wouldn't ring. I tried accessing the software to reboot it but couldn't find the option for that and ended up having to crawl under the desk to get to the base to unplug it.

On a similar note, how is the reliability of these. Mine is less than 2 weeks old and if it needs power cycling often I'm going to have to find a better place for the base :/

SteveM writes...

On a similar note, how is the reliability of these. Mine is less than 2 weeks old and if it needs power cycling often I'm going to have to find a better place for the base :/

I have never rebooted it. I did apply the latest firmware when it showed up.

kko writes...

I got my S685 IP (base station and four handsets) from hip-ip.com, a UK company

Forgive my questions – I am new to this VOIP thing but am interested in Siemens solution.

If your phone came from the UK did you have any issues with the phone line plug (PSTN)? Was a converter required? From memory the phone plugs in the UK were slightly different to OZ but I could be dreaming.

If someone calls you on your existing Telstra (non-SIP) number will this phone connect to the inbound call (and ring)?

Are there any limitation on the usage of the existing telstra line for outbound calls?

Cheers.

Lambretta writes...

If someone calls you on your existing Telstra (non-SIP) number will this phone connect to the inbound call (and ring)?
Yes

Are there any limitation on the usage of the existing telstra line for outbound calls?
No, you dial a prefix to send the call via PSTN then it works like any phone.

Lambretta writes...

any issues with the phone line plug (PSTN)? Was a converter required?

I just used a spare replacement line from an old phone. Or you can cut off the UK plug and crimp a new one on the end.

If someone calls you on your existing Telstra (non-SIP) number will this phone connect to the inbound call (and ring)?

Are there any limitation on the usage of the existing telstra line for outbound calls?

In-Out call control is really easy to configure.
+ set default provider (PSTN, or any of 6 VoIP providers)
+ override the default with suffix of #n (n=0-6, 9) to indciate which PSTN/VoIP provider
+ for each handset, turn on or off which PSTN/VoIP providers can call in or out
+ basic but efective dial plans, based on prefix, directing to any of the available PSTN/VoIP providers

the only negative thing, that is turning me off this phone, is the inability to set more complex dial plans.

ideally i would like to use SIP1 for local/national, and SIP2 for mobile calls.

i realise you can dial a prefix to use SIP2, and i'll remember this, but my wife wont...

does anyone think this might be addressed in a future firmware update? :)

thanks!

SteveM writes...

Mine was registered, and I could call out, but it wouldn't ring.

SteveM,

I had the same problem – could ring out but not in. Same problem on 2 separately registered accounts. Curiously, when I tried to call out on one line and in on the other (I have 2 handsets and two voip accounts) I heard nothing on the line, the phone did not ring, but it immediately put a missed call in the call register.

I upgraded firmware about a week ago – until then it never missed a beat in about 6 months. I am suspicious of new firmware.

What version do you have – I have updated to latest version:

021230000000 / 041.00

Very unhappy if this update has made it unstable as it has been a great unit to use :-(

['What version do you have – I have updated to latest version:

021230000000 / 041.00

']

That's what I have, I updated it when I installed it. Was your problem both accounts at once, or one account at a time. I only have one account set up as it is to an asterisk server – I haven't even connected the PSTN to it

SteveM writes...

Was your problem both accounts at once, or one account at a time.

Both at once. Calling in on PSTN worked fine. Power cycle fixed the problem.

I bought my C470Ip this morning from JB-Hifi. Did the same as the others, showed the sales guy a printed page from this thread, he looked in his computer and gave it to me for $139. I am very happy about this price! They do not have any additional handsets in stock, but he said he could order them in (he wasn't to sure dough), which probably means no good deal... If anyone knows where to get the additional sets for cheap in AU please let us know!

Went home and hooked it up to my (rather old) asterisk box. Had one small problem: it would not work when plugged directly into my ADSL modem (Thompson 585), even the link led went on on the router, plugged it into my switch and since then I had no problems, everything is working. No MWI however, my asterisk is to old to support subscribemwi=yes, so maybe in the near future I update my whole server to the latest *.

So far I am very happy, I used to have an old dect phone hooked up to a ATA and when it died I got a Zyxel2000 Wifi Phone which was never any good and it has since died anyway. So I hope with the C470 I can finally use the telephone as I used to do before VOIP... and this (hopefully) makes the wife happy!

theguru1 writes...

Hi,

SO what is the difference, if you know, between the 475ip & the 685? Apart from 25 pound.

Any one?

Lee writes...

ideally i would like to use SIP1 for local/national, and SIP2 for mobile calls.

Very easy to set this up in the dial plan. Dial plan will match anything starting with 04 and direct to whichever SIP account you want. I run two SIP accounts, and have set my dial plan to direct local/national calls dialled with an STD prefix (02, 03 etc) to one account, and ones dialled without (9..., 8... etc) to another. To make it 'wife friendly' I have entered our most commonly dialed numbers into the address book with or without prefixes as needed to direct them to the right SIP account.

The dial plan (such as it is) works by matching the first digits in the dialled number, and works fine for most simple operations like this.

What it won't do is more complicated things like drop prefixes, substitute numbers, etc.

metropolis writes...

Very easy to set this up in the dial plan

oh great, will have to really consider this phone then! thanks!

Apart from the higher resolution screen on the S685IP, can anyone tell me what the difference is between it and the C475IP? Please

theguru1 writes...

can anyone tell me what the difference is between it and the C475IP?

As I am aware, the range differences are:
+ C470-IP Basic.
+ C475-IP + AM (Answering machine)
+ S675-IP + AM, HDSP (High Definition Sound) (may not be an English version speech prompts??)
+ S685-IP + AM, HDSP, Bluetooth

Bluetooth allows easy send of contact detail from mobile to a handset. You can then send from one handset to another. Much easier than the import method.

HDSP is g.722 codec. Not many current VoIP providers currently support it. Gigaset users should be able to call each other! Must be supported both ends and have sufficient bandwidth.

I have the 685-IP and we are very happy with it.

See also /forum-replies.cfm?t=959866

metropolis writes...

The dial plan (such as it is) works by matching the first digits in the dialled number, and works fine for most simple operations

metropolis is right. The dial plans are VERY EASY to enter and manage.

Compared to the struggles I and others have had with Linksys ATA 3102 and similar, these are a walk in the park.

Timed Out writes...

I have the 685-IP and we are very happy with it.

Hi,

Thanks for the info.

Where did you buy your S685 and did you pay VAT? I understand it should be VAT exempt.

Timed Out writes...

As I am aware, the range differences are:
+ C470-IP Basic.
+ C475-IP + AM (Answering machine)
+ S675-IP + AM, HDSP (High Definition Sound) (may not be an English version speech prompts??)
+ S685-IP + AM, HDSP, Bluetooth

Bluetooth allows easy send of contact detail from mobile to a handset. You can then send from one handset to another. Much easier than the import method.

HDSP is g.722 codec. Not many current VoIP providers currently support it. Gigaset users should be able to call each other! Must be supported both ends and have sufficient bandwidth.

I have the 685-IP and we are very happy with it.

See also /forum-replies.cfm?t=959866

Just to add to the above short n concise info

The Base Stations of the C470 IP and C475 IP support HDSP but the Handset(C47H i think) supplied with the package does not.

If you use a HDSP compatible Handset, i believe S67H and SL37H are, then you can use it for HDSP conversations as well.

Cheers

Can you use a variety of DECT phones to connect to your Siemens?

And how do you go about it?

sorry left out quote

eug k writes...

metropolis writes...

I have even used this on an older Siemens DECT GAP handset (can't recall model) which was supplied with a bog stock PSTN base, which was able to register to the 470IP base and work fine.

With a non-voip DECT GAP handset, how do you select the outgoing line (pstn or sip account) before you dial?

Hi, I'm having a problem with my C470IP phone that I'm wondering if someone could help me with? When I use the standard phone line (not VOIP), I get about 5 seconds of interference within the first 30 seconds whenever I make or receive a call – has anyone else found this problem? I have no idea what the issue is, although I do run broadband (wireless router) through the same phone number – could it be this that is causing the problem? Any help would be much appreciated!
Thanks, Rachel

Apache_AU writes...

Can you use a variety of DECT phones to connect to your Siemens?

And how do you go about it?

I have used a Doro 530 and a $10 Dick Smith special DECT handset with my Siemens S450IP and C470IP systems.

You put the Siemens base station in "registration mode" by pressing the blue button for a second or so, then follow the instructions for the handset to register it to the Siemens base – the procedure differs between different types of handsets. The PIN number required is the PIN of the base (0000 by default).

To select PSTN from a non Siemens handset you just add #0 to the dialled number
(and #0 -#6 to select a SIP account). You can set the default account from the Siemens GUI to be PSTN or 1 of the (up to ) 6 SIP accounts.

Of course with the generic DECT GAP handsets you don't get the special Siemens features such as Phone directory – very useful feature (or SMS/Mail/Weather/RSS etc which is not an issue for me as they are just of novelty value.

Trevor

I have a C470ip and currently have have it working through my Netgear WNR834B (running DD-WRT V24Sp1). I have it setup to use 2 VOIP providers from the UK. I have forwarded ports 5060 and 5004 to the C470 on the router.

I have no issues with calls at all with this.

I have swiched from Bigpond to Naked DSL with iinet and the broadband service is working fine. Last night they activated my VOIP service and I am having some issues with it.

I can make calls without a problem but when I try to ring my number it goes straight to voicemail. I spoke with iinet and they said to chabge the RTP setting to 5100. i did this on the C470 and changed my forward port on the router from 5004 to 5100.

This worked initially but this morning when I tried to call my number the phone rang but I couldn't here the caller. THe other times again it went straight to voicemail.

Can anyone offer any help with this?

Thanks in advance,

James.

You need to forward the port RANGE from 5004 to 5100, not just the one port number 5100.

boofh writes...

You need to forward the port RANGE from 5004 to 5100, not just the one port number 5100

Thank you for this. I will forward that range on my router tonight. What about the setting for RTP on the C470? Should I change that back to 5004 or leave it on 5100?

Cheers,
James

I don't have access to mine to look at the moment, but I never set that, just leave it as default "out of the box".

From default factory, you should only have to set the registration info and then "sometimes" do some port forwarding on the router (depending on brands / type), that's it.

I will let you know the outcome after I test it tonight.

Thank you for your help with this.

James

I couldn't wait until tonight so called home and talked my wife through the changes.

C470 back to RTP 5004

and port forwarding 5004-5100 to the CP470.

Just tested it and works fine.

I performed an automated firmware update on this unit and suddenly the voice quality dramatically degraded. Didn't matter which codec was chosen, audibility of the call from my end fluctuated from microburst of cutouts.

The other end had no issues hearing me clearly, problem was I've had to ask the other caller to repeat themselves when constant voice cutouts were occuring.

This did not at all happen on the ORIGINAL firmware loaded. Anybody have gotten a hold of older firmwares. I've emailed siemens and have yet to answer after 2 days from their email receipt confirmation.

Anybody else having issues with the new firmware?

-P-

OK i have just picked up a C470IP from JB in Narre Warren. after a quick discussion got it for $129.
now my only problem is trying to get my billion743ge to work with it.

All good now, seems to be working.

The C470IP seems to have been discontinued from all the online stores, and they aren't promoting a replacement. My first one is working a treat and I want to buy two more, does anyone have a suggestion for an alternative? I know about the SNOM but it's still a bit expensive.

I can try and drive around (or ring I guess) to JB stores and see if I can find any stock, but I don't really have time for that.

You want the whole thing or just extra handsets? I've been using Doro 760x handsets with my c470 base and I prefer them. You don't get all the 'extras', but its a nice sort of muck resistant handset which is good with small kids ;)

Freshtel seem to still be selling them.....

Hi Tankman,

Yes, Freshtel have plenty of them and an online store.

Regards,

– David.
Freshtel Pty Ltd

lee writes...

OK i have just picked up a C470IP from JB in Narre Warren. after a quick discussion got it for $129.

Took a copy of this page with your pricing info to our local JB. The phone is listed in the cata;gue at the front door at $199.00 with $100 credit if sign up with Freshtel.

When I finally found a salesperson and found the box (hidden at end of a row) I showed him the page from WP (which he hadn't heard of) and he agreed straight away to the $129. Thinking that he responded without any hestitation, I asked if he could do better. He then consulted the computer and responded with $115.00. Deal done. Unfortunately, last one in store, but he did say that more would be coming in.

Now to get it working. Unfortunately, I dont have a fixed phone and ethernet connection together. I have got it to make a pstn call.

theguru1 writes...

Unfortunately, I dont have a fixed phone and ethernet connection together. I have got it to make a pstn call.

Since you already have Wifi, you can use one of these wireless bridges:
https://oztechnologies.com/detail.asp?product=WBP54G
https://oztechnologies.com/detail.asp?product=PA100
plug it into your C470IP ethernet port and put the C470IP base station where your PSTN line is.

Now you have both VoIP and PSTN together.

We use quite a few of these on SPA941 VoIP phones to make them wireless and there is no different in call quality between hardwired and wireless phones.

Probably just end up running another ethernet cable under the house. Haven't been under for a while, the height is fairly low and I end up having to crawl on my back into position, not enough room to turn over.

Well after reading this thread i went to the local JB and asked what the best they could do was.

After checking they came back with $110.

the sales guy needed to go though 5 boxes before finding one with a basestation power supply but hey i have it now.

Now to work out if my telstra F5100 handset will work.

Setup help needed.
Can get the phone registered with MNF, can make calls however can not receive calls . Have tried port forwarding, but no difference. Using a Netgear WGR613 locked, and I guess this is the problem as I can't turn off voip on the device.

Someone?

theguru1 writes...

Using a Netgear WGR613 locked

Are you trying to use the same SIP account as the one on the 613? You cannot have a MNF SIP account registered on more than one device at a time. Your best bet is to either switch off the 613 (not possible if your 613 is your router) or purchase a one-line BYO account to get a different SIP account (which will come with a different DID).

CableRider writes...

purchase a one-line BYO account to get a different SIP account (which will come with a different DID).

Would it then be possible to combine that account with my existing neosaver account?

With this phone, do you have to select whether the call is going out by the PSTN or VoIP on each call, or can it be set to always use VoIP by default (I have no PSTN line)?

Also, I'm considering getting this in addition to my ATA, is there anything I would need to do to run what would essentially be two ATAs from the one router (eg can they both use port 5060 concurrently?).

Cheers

theguru1 writes...

Would it then be possible to combine that account with my existing neosaver account?

No, I'm afraid not, so you might have to convert the existing NS to a WPS and use the new one as a NS account instead. You would also want to ask them to transfer the 613's DID across to the new NS account too.

I ended up getting one of these units too and have one issue.

Periodically my cable service may drop out for a minute. Following the cable modem's successful reconnection, the phone fails to re-register to MNF. My ATA registered to Pennytel re-registers. The registration time is set to 180 seconds (default). If I pull the power from this for a second and plug back in, it registers again immediately.

Any thoughts?

Nerf writes...

Periodically my cable service may drop out for a minute

I would contact the provider for a start. It shouldn't drop out. I'm using the Siemens with adsl atm, but haevn't had an issue with drop out.

It is an issue and Bigpond refuse to engage in any further discussion on it, simply referring me to Gizmo to pay for tech support. I did have a tech come out, open his laptop and declare that as it was opening webpages fine at that time, there was no problem with my connection (even though a tech on phone support said my upload voltage was far too low...).

But back to the Siemens phone, I don't understand why if the re-register period is set to 180 seconds and say I power-cycle my modem, surely after it has reconnected the phone should at some point within the next few minutes be able to register again with MNF?

Hi all, just want to add my 2c for those interested in these phone sets; I ordered a c475 from Hip-IP in the UK, arrived about 2 weeks later in a bubble wrap bag. Phone is clear sounding and easy to set up, although I still have to work some stuff out. Much better than the old ata to phone setup. One issue was when I updated firmware the voice quality went rubbish, but now has cleared and I'm not exactly sure what fixed it, perhaps the codec settings I think. Just Whim any questions, but I haven't tried many of the features yet.

I found that the firmware update didn't seem to complete the first time, so I re-ran the update again.

Hi can anyone help me?

I have a GigasetC470Ip, last week I signed up with Gotalk and can't get it to work. I have started a thread here ...

/forum-replies.cfm?t=1050053#bottom

any advice appreciated

Thanks

Has anyone been using gotalk on their Gigaset?

Apache_AU..

Check my reply on your thread...

/forum-replies.cfm?t=1050053

cheers.

I got one of these phones this week. After a little setup it was running fine.

It took a couple of goes to get it to download the new firmware, in fact I would say that it would only allow me to upgrade the firmware via the menus on the handset, the web interface did not work

I like the fact that I can register my old panasonic dect handset with it and then set that up as either skids extension or as a separate billed phone for use by friends and family in the extension. To do that with ata I would need two atas which would cost more than this unit

Thanks to everyone who has contributed to this thread

About to purchase one of these C470IP's they sound just too temping.

There are a few places to purchase here and OS but prices are high (considering others here have quoted around $120-130 from JB)

Best price ATM is JB online at $199.00 plus post

Anything better around ???

BTW JB say their phones are sold with freshtel label but you are not compelled to take a contract with Freshtel, Anyone know for sure ?

Bthree writes...

Anything better around ???
JB instore.

TW JB say their phones are sold with freshtel label but you are not compelled to take a contract with Freshtel, Anyone know for sure ?
There are 2 types in JB, one with Freshtel sticker with free amount of calls, and one without freshtel label.

Both are NOT locked to freshel, and there is no "contract".

Bthree writes...

Anything better around ???

I bought one this week from JB HiFI Indooroopilly (QLD), I too could not get as low as the figures mentioned on this thread, but I was happy enough with $150, my price point was 140, but as my other ata had died I needed one asap, so I was happy enough.

Mine had a freshtel sticker on it, but as mentioned in this thread, it is not locked....

I am new to voip on mynetfone with spa3102, going okay but quality a bit of an issue. i like the sound of the c475 for is answering machine. due to it not being in aus would probably go the c470 my question is can you hook a standard answering machine in front of the c470. (the phone line i am talking about)

Cheers

Mulver writes...

can you hook a standard answering machine in front of the c470

Don't see why not, it is still the pstn line. It won't of course pick up voip calls, but the vsp message service can answer those.

I figured as much. not yet using voip for incoming calls as we cant get adsl2 so have to have a fixed phone line might as well use it.
Cheers

TEDS have the Siemens A260 Gigaset cordless phone for sale at $49.95.

Anyone used these and are they usefull as additional handsets to use in conjunction with the C470ip?

Hi! can you please help me? I have a c470ip and a big problem. When I call someone the other party can hear me very well but I cannot clearly listen to him. the voice is interrupted and I cannot understand anything. My phone is connected to a linksys router.
download speed: 2164 kbps
upload speed: 534 kbps

What are your upload/download speeds?

Try a couple of tests and let us know the result.

http://www.testyourvoip.com/

http://www.speedtest.net/

download speed: 2164 kbps
upload speed: 534 kbps

Those speeds should be OK – just make sure that you are not uploading/downloading while you are using VOIP.

Have you tried port forwarding 5060?

Also set the web configuration >Telephony>Audio> Optimized for low bandwidth

and see if that makes a difference.

theguru1 writes...

TEDS have the Siemens A260 Gigaset cordless phone for sale at $49.95.

Anyone used these and are they usefull as additional handsets to use in conjunction with the C470ip?

According to the Siemens Gigaset compatibility database, the handset (A26) can do everything that the C47H handset can, except for:

- Configuring the Base Station
- Caller List
- Message Waiting Indication
- SMS

So feature wise, they're ok, sort of...

But I haven't tried one yet – I'm still hoping that I can get the C47H handset.

That made a difference. But still I hear voice interrupted. Are their any special settings for my router?

how to do this?

I assume the TED's supplied A260 is a PSTN base station plus A26 handset.

One potential problem will be that the A26 will come registered to its own base. You can certainly register the A26 handset to the C470IP base (part of the DECT spec.), but you will have to charge the A26 in the A260 base and you may find that it "prefers" that base over the C470IP base.

I have a similar situation here – A C470IP base with several C47/S45/C45 phones registered to it and S450 Analog base (no PSTN connection, only using it for charging) with S45 phone.

Even thought the S45 is not registered to the S450 Analog base (I think – difficult to check) it is in close proximity to the (unused) S450 Analog base transmitter and sometime loses its registration to the C470IP base which is several metres away.

I may have to "hack" into the S450 Analog base to totally disable all but the charging function if I get too annoyed by the issue.

Trevor

Hi
I also have a 470IP and was wondering...does it support CLI on PSTN calls? i dont seem to b receiving any CLI ?

Secondly, i am using SIP via voipbuster and it works well but when trying to use another provide www.12voip.com , although it registers succesfully, it does not make calls – it says NO IP CONNECTION
any advice?

thanks

Hi zforum69,

Very nice review. I just got a set myself and the build quality is quite nice. Unfortunately the the adaptor for the base wasn't in the package and the salesman said to ask Freshtel to send the adaptors.

May I ask what is the input voltage and current of the base? Also is the plug +-+ or -+-?

Cheers :)

I got the GIGASET working with 12voip now...for some reason when dialling i have to dial #1 or #2 etc to force it to use the SIP profile on which 12voip is loaded.

ANYONE OUT THERE : I am not receiving CLI on normal analogue calls whereas my other fixed telephones on the same line are receiving CLI....
what to do?

FirestormX,

The plugpack for my C470IP is rated at 6.5 VDC 600mA

The "centre" pin is positive.

The no load voltage I measured was 7.65 volts. (I cannot measure the on load voltage without opening the C470IP case)

Judging by the weight it is a switchmode regulated plug pack (as opposed to a traditional transformer/rectifier type that usually have a no load voltage up to twice the voltage on load)

The model is a SALOM IP40

Trevor

Thank you very much TrevorB.

Cheers mate :)

FirestormX writes...

May I ask what is the input voltage and current of the base? Also is the plug +-+

The plug is a NON standard size ..... well not a 2.1or 2.5mm DC connector
something a little larger which I have been unable to identify ??

Any of you that didn't get plug packs, it is JBHifi's responsibility to supply those to you by law. They can ask you to go to Freshtel, but if you insist JB must support you directly or give you a refund.

We had our missing plug pack couriered to us on JB's courier (but when I bought the phone the best price was $169)

Thanks for the info boofh :)

I'll see what Freshtel says tomorrow and go from there. I wonder why the plugpacks didn't make it into packages.

Cheers :)

Edit:

Cheers BThree. If someone can give me the exact dimensions of the DC connector it would be great. :)

FirestormX writes...

....I wonder why the plugpacks didn't make it into packages......

All other Siemens phones for sale in Australia are not IP phones and have the base and the handset together, so only one plugpack is required per handset. Perhaps whoever added the Australian plugpacks made an error and didn't realise that the C470IP requires 2 plugpacks whereas the C470 requires 1, even though they use the same handset.

For an example of how this confusion can occur www.ryda.com.au/Siemens-C470IP-Voip-Cordless-Phone-p/c470ip.htm still shows the incorrect photo, even though I advised them of their error 6-8 months ago.

anyone there who can help me regarding the CLI? im not receiving it at all on analogue calls...

FirestormX writes...

If someone can give me the exact dimensions of the DC connector it would be great.

DC style connector Plug outside dimension is 5.5mm same as 2.1 and 2.5mm but the centre pin appears to be 2.7mm ( a little larger ) !

anyone there who can help me regarding the CLI? im not receiving it at all on analogue calls...

Bthree writes...

DC style connector Plug outside dimension is 5.5mm same as 2.1 and 2.5mm but the centre pin appears to be 2.7mm ( a little larger ) !

Thanks Bthree.:) Unfortunately it looks like these things are going to be hard to replace if something breaks.

mams writes...

anyone there who can help me regarding the CLI? im not receiving it at all on analogue calls...

Did the CLI work before adding the C470IP? Works fine for me.

CLI is active on my line and works on my normal fixed telephone. Both are connected to the same line.
The C470IP for some reason does not display any CLI at all.
(simply put, my fixed phone shows a CLI and my C470IP shows no number)

I cannot understand why....CLI is not a function that has to be activated on the handset as far as i understand...it just needs to be active from your analogue telecoms provider...or am i missing something?

mams writes...

The C470IP for some reason does not display any CLI at all.
Have you firmware upgraded the C470IP ?

yes, that was the 1st thing i did....i have sent a message to siemens asking them for a solution but no reply from them as yet....
what else can i try?

i am now receiving calls via my DID forwarded to SIP and that is giving me CLI....its only the analogue calls that give no CLI....anyone out there???please help!

Could be faulty or it could be low CLI signal levels you will have to do some basic testing first. Testing using another phone isn't always 100% proof the signal levels are OK either as some phones will trigger on a very low CLI signal level. Having a high sensitivity to CLI signals is not ideal either as this can cause false alarms on the display ie showing text just because you whistled or something beeped in the office and it is picked up by the handset mic triggering the CLI circuit (Yes I have actually seen this happen).

First make sure there are no other phones plugged into the line (some faxes/phones for some reason pull the CLI signal down pretty hard) then ring the phone from a mobile. If nothing is showing and another phone works on the same point try plugging the handset into another telephone line and see if it works. If the C470IP works on another line it means you have low CLI signal levels so you will need to talk to your provider.

Thanks for the troubleshooting advice...
I now tried – as u suggested – ringing the c470IP without any other telephones connected to the line and still nothing....

Also tried another line and still nothing....

Can it be any settings related problem? (i dont see any settings for CLI on any phone)

That leaves it probably to THE UNIT BEING FAULTY?? hope there is something else that sort its out as i bought it in South Africa and am living in Saudi Arabia...sending it back will be problemsome..

Australia uses Bellcore/Telcordia GR-30 CORE CLASS (Custom Local Area Signaling Services) standard*
Anyone from Australia would be using this standard and so their C470IP CLI experience may not be relevant to you in Saudi Arabia.

South Africa uses ETSI ETS 300 648, ETS 300 659-1 CLIP (Calling Line Identity Presentation) FSK standard*
Saudi Arabia uses ETSI ETS 300 648, ETS 300 659-1 CLIP (Calling Line Identity Presentation) DTMF standard variant 1*

Here is a page with a C475IP for sale including a warning that this Belgian approved model uses FSK CLIP and is not compatible with DTMF CLIP.
www.voipsolutions.be/voip-wireless-dect/siemens-gigaset-c475-ip-voip-phone.html

A C470IP purchased in South Africa would possibly also only work with FSK and not DTMF. A DTMF to FSK converter may fix this. For example I have seen this one suitable for Siemens phones in the Netherlands:
www.telec.nl/producten/p55224/fsk_naar_dtmf_of_dtmf_naar_fsk_omzetter.html

You would have to check for the correct converter yourself, other ones may also work but I have no knowledge or experience in this area.

This may or may not apply to your C470IP as there are different versions for different markets. It is just a possibility for your consideration and may be totally incorrect.

*Standards from www.soft-switch.org/spandsp-doc/adsi_page.html

Thanks for the detailed reply!
at least i know what is the most probable problem and can start tackling it...
Thanks a million.
ill start my search for an adaptor here and in South Africa for a DTMF to FSK converter

I was going to suggest an incompatible CLI system, but Kasi has said it all.

Hi, I am new in this forum. It seems that you have good knowledge with VoIP Gigaset c470ip. If is possible could you please send me an email (fcamargo100@hotmail.com), and explain how to set up this phone connected to my computer with Ethernet line?

I tried already but the phone doesn’t detect any IP address and don’t know how to do it.

I don't have any router, only a USB wireless Modem from 3G.

Also if you have used already pennytel, I need the configuration please.

I will appreciate very much your help in this matter,

Regards

Phillip.

valle writes...

and explain how to set up this phone connected to my computer with Ethernet line?

Your first issue is not really with the phone or VoIP, it is networking. You have to use Google and spend some time learning about "ICS – Internet Connection Sharing" for the operating system you are using on your PC.

You have to set your PC up to "share" the internet so that other devices can connect through it.

You would be much better off buying a router and plugging your 3G modem (if compatible) into the router and using the router to share your internet and protect your PC's.

A suitable router is the Draytek 2910 or similar.

Hello all,

after reading through this thread I went out (yesterday) to JB at Parramatta Westfield on Level 5 to see if they have stock. They do have about 7-10 boxes at marked $199 each. Some with Freshtel label and some without. I asked a friendly sales guy for the best price. He calculated and said $150. I didn't answer straight away. Next he asked if I had something else in mind. I said I was reading at whirlpool about this phone and that people could buy it at other JB for $110. He replied that he has not heard of whirlpool, but would take my word and offered me to have it for $110. Then I mentioned that some boxes seem to miss one power pack. I asked if he wouldn't mind to check the content. He opened the still sealed box and checked. Lucky me, two power packs with everything else inside. I said thank you very much and paid the $110. I am very happy with my buy.
Back at home I removed my old phone (Panasonic) setup. Plugged in the C470IP as per guide. Disabled my VoIP on the Belkin router. Setup NAT and open port 5060 and 5004. Setup my VoIP account via web page for the C470IP. It was relatively easy to do.
I followed the guide to charge the phone first. Today I made my first VoIP call, and I happy with the result.
Thanks a lot to everyone and their input to this thread. You made my day.

Cheers,
William

Hello again,

I would like to get a headset for the Siemens C47H phone. Maybe a Plantronics MX150 or a headband style one.
I am interested what other people are using or would recommend to buy.

Cheers,
William

weigenmann writes...

I would like to get a headset for the Siemens C47H phone. Maybe a Plantronics MX150 or a headband style one.

The handsfree works quite well on these phones.

I agree, the handsfree works very well.

We use a planstronics headset, but it's a few years old and I have no idea of the part no, it is a headband type with earpiece on only one side and a boom mic, and a little volume control in the middle of the cable, the quality of the calls using it is outstanding.

It's really nice to be cordless with VoIP handset headset, being able to move around the office.

boofh writes...

It's really nice to be cordless with VoIP handset, being able to move around the office.

I constantly get told off for not staying still when I take calls. I walk around all over the place.

Thanks to everyone for their input.
I ended up buying a Plantronics M214C headset (over-the-head style).
Control of talking and listening volume.

-= Mute with one-touch =-

That completes my new phone system for now.

Picked up my phone to see the red missed call light flashing. After cancelling that the screen came up with a "Firmware Update Available" text.

Brave me immediately goes settings/base/Software Update and .......

Nothing..... then after about 20min the handset came back on line, phew

Firmware Version now 021390000000 / 041.00

Was 021230000000 / 041.00 can't see any difference !

Good luck all

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

Bthree writes...

Nothing..... then after about 20min the handset came back on line, phew

It always takes a long time.

Can this phone be connected directly to a lan port of my laptop whilst connected to a hspa 3G connection to make calls without the need to purchase an ata and router as I dont need an wireless connectivity and I dont have any adsl or lanline phones

Should work like that. You would need to leave you laptop on all the time you want to use the phone of course. The alternative is a softphone on your laptop.

I currently have a voip modem, with a Uniden cordless phone which is and was a problem ever since I bought it. Now I am looking for a better phone to replace it with. I have come across the Siemens via this forum, and I need to know if instead of buying yet another ordinary cordless phone and just hope that it is better than the Uniden, I can get the Siemens, and use it via a spare ethernet port on my voip modem, and cancel the voip settings in the modem. IE, treat my voip modem as just a modem, and use the Siemens instead to do all the voip work. I am sure that this would work, but I would like clarification from somebody here that knows for sure. Thanks.

What VoIP modem do you have?

There are some reports of Billion modems 'eating' SIP packets from the VSP even when the VoIP function is disabled – YMMV.

The cheapest way to check would be to install a SIP softphone (eg. X-Lite) on your PC – if you can make & receive calls, then the Siemens should be fine.

It is a Draytek Vigor 2700 V . Refer here /forum-replies.cfm?t=1030215 As you can see if you can bear reading through it all, I am over all the ins and outs of it all. In the end all my problems were the fault of a chinese made junk handset, ie, Uniden. So what I am now wanting to do is to bypass the need to try another chinese made junk handset with a different brand name and do something different using hopefully a better quality product.

Will the Gigaset M35 work with my C470IP so that I can also use it for Skype calls?
or alternatively does the C470ip handset support multibase so i can connect the handset to another base which is M35 compatible as well as being connected to the C470ip base thus allowing Skype calls as well?

EDIT: I checked and the 470IP handset is not multibase compatible – the 675IP and 685Ip handsets are multibase compatible = Question: I f i buy a 675 handset will that work via the 470 base coupled with a M35 compatible base?
BEST option would be if the M35 is compatible with the 470 base....that question remains...anyone?

Mams, where did you find the info on the 675 IP handset being multibase compatible? I've been looking all over for it.

Has anyone tried to make two c470IP or S675IP work together? I've been using a S675 IP for a year and am very happy with it (never dared to change firmware tho'.....) But I use it solo (on both PST and 1 single SIP account) and have no experience working with several handsets.

The tiny company I work for wants to go voIP. But we need 4 simultaneous lines (6 sip accounts) because we're constantly on the phone and that would mean using 2 bases. One important requirements is to be able to have all handsets ring if one direct line (SIP account) is not picked up after 5 rings. Would it be possible or would only 2 handsets ring? Another one would be able to pass on the call to another handset from the list of handsets (like the internal button does on my S675IP). Would all the handsets be listed or only those attached to one base?

Thanks in advance for your feedback/advice.

check http://gigaset.com/shc/0,1935,hq_en_0_152411_rArNrNrNrN_variation%253A-5_pageType%253ATechnical%2Bdata_imagePos%253A0,00.html#content

under TECHNICAL SPEC it says:
Multibase Multicell for up to 4 base stations

Marie-Jo Jones writes...

But we need 4 simultaneous lines (6 sip accounts)

Me thinks you need a more elegant solution

Even the S675IP can only handle 2 voip and one PSTN line regardless of the number of base repeaters (max 4)

Thanks for the info. Called a Siemens distributor today and he confirmed that tho the S675IP or S685IP are multicell (4 bases), these bases function in parallel only. Hence even if you can register their respective handsets on 2 or more bases, you have to switch manually from one to the other if the one you're on is fully occupied. If an outside call comes to a base which has no simultaneous communications left, it will get a busy signal and will not be transferred to the other base. Moreover you can only transfer calls to the handsets active on the same base at that time (you can of course dial the transfer number manually or pick up the number from the contact list).
Pity 'cos I like this phone... Only regret that the settings are not totally individualized (list of calls doesn't give the list of made calls, it only lists missed calls or all calls, ringtones are identical for all sip accounts while it would be handy to have different tones to identify the line, i.e. like private or business, before picking it up the phone, diverted calls are set on missed calls or all calls without settings for each occurrence).
For the rest it suits me well.
Thanks again for your input and wish you a merry Xmas

Thanks, agree that Gigaset multibase is not a solution for a small company 'tho it might be OK for a family using a base for private use and a base for business use.
But what would you suggest as an alternative.
The only phone with 3 simultaneous voIP lines I could see is the SnomM3. But it doesn't have a PST line which I think is a drawback if network goes down. And it's quite a bit more expensive.
Don't want to run an Asterisk server 'cos I don't have the time to delve into this and I'm often on the road. So if something goes wrong my colleagues would be in trouble.

Hey All,

I thought I should tell of my experience of trying to get my recent purchased C470IP working. Basically, the unit was behaving very badly, and not able to register to my VoIP provider (NodePhone), nor even gigaset.net. Even the automatic firmware upgrade failed with a 404 transmission error.

After manually upgrading the unit to the current firmware and reading on the gigaset-blog about issues around NAT I discovered the following:

(1) my modem Billion 7300-A with new firmware version 1.36-698_sso_billion
(2) NodePhone Siemens C470-IP upgraded to the last firmware
(3) NodePhone ATA SPA-3102 (to be replaced by the C470-IP)

The issue seem to be with how NAT and SIP / RTP is being handled by the 7300, Downgrading to v1.33 fixed the issue.

/forum-replies.cfm?t=983165&p=2#r33

I hope it saving someone some millage

Nik

In case anyone has the problem i had with no CLI on analogue calls due to FSK-DTMF compatibility issues, i have receieved a working solution from Siemens in Dubai.
The solution is:

Menu -> Settings -> Base -> 9 4 7 6 2 0 0 1 -> 09447 01636 13979 -> OK

mams writes...

Siemens in Dubai.

How on earth do you end up getting support from there?

Z

I got someone back in South Africa to call Siemens in South Africa and explain my problem. They then referred me to Siemens in Dubai as the IP models are not supported nor available from Siemens Saudi Arabia

Just letting anyone interested know that i ordered a couple of C475IP units and extra handsets from the UK and have just received them and they work well, very happy with the units so far.

The C475IP has the integrated answering machine unlike the locally available C470IP and extra handsets are readily available.

I ordered two C475IP units, one with two handsets and the other one with 3 handsets.

Unfortunately the company i ordered through would not send internationally or allow an international credit card to be used, so i had to get a friend in the UK to buy them for me and ship them to me but i am sure there are probably other companies that will send international.

I've used Korjo UK->AUS plug adaptors for the power packs (needed 7 in total!) ordered from the Korjo website although BIG W also sell 'Jackson' brand adaptors but they are a little bulky and a couple of $ more than the Korjo ones.

Anyway if someone ends up ordering the C475IP, the UK is a viable place to get them from.

does anyone know if I can pair an older Siemens c340 handset to the c470IP base?

i've been searching for days and can't find any helpful information!

edit- the c340 is DECT/GAP compatible. I think this means that it would work?

Lee writes...

does anyone know if I can pair an older Siemens c340 handset to the c470IP base?

I am associating an older phone and this is what I can do. The post is an extract from a whim to Z re getting 3-way calling working.

url www.amazon.co.uk/Diverse...ne/dp/B­ 000060OBI. I bought 4 of these (re-badged Siemens Gigasets) in the UK about 5 years ago. They can do internal transfer calls via the "INT" key, and also have that nice facility whereby you can copy the directory from handset to another. For further info on the phone this takes you to the manual www.viking-direct.co.uk/...010executive.pdf. 2MB download.

I have associated them with the C470IP and set myself 3 objectives.

1. Internal calling. On the Diverse if a call came in I could transfer it to another handset using "INT" easily because the handsets are named. However, with the C470IP I can't see the names on the Diverse handsets as it just displays "Base 1". Pressing "INT" shows me nothing although I have named the handsets via the web interface.
The solution was rather easy. I added the handset names to the external address-book e.g. name "3-lounge" number to dial "3". Now I press "INT" and use the external directory find the handset I want to transfer the call to.

2. Transfer the directory of Diverse handsets to each other and the C47H. This didn't work using a Diverse handset.
The solution was to use the web interface to copy the directory of a Diverse handset to my PC and then copy it to the C47H. In fact, it went one better in that I opened the .vcf file as a .txt, added new entries manually and the saved as .vcf.

So far so good and I'm thinking those that want extra handsets just need to buy cheap handsets locally provided they have the "INT" key and can copy directories between handsets.

3. I have a teenage daughter that every now and then has requested the facility to call two friends at once. Given the cheapness of VOIP calls and not having to pay for the 'special feature' from Optus I am happy to acquiesce. I know I can do it using the C47H but that's my handset and if I allow her to use it at all I have this feeling it will become permanently stationed in her room.
I'm looking for a way for the Diverse handsets to set-up a 3-way call. I'm hoping this will involve pressing "R" and some dtmf tones? Any ideas?

mams writes...

In case anyone has the problem i had with no CLI on analogue calls due to FSK-DTMF compatibility issues, i have receieved a working solution from Siemens in Dubai.

The solution is:

Menu -> Settings -> Base -> 9 4 7 6 2 0 0 1 -> 09447 01636 13979 -> OK

Hi Mams does not seem to work here in Aus for me? Any other ideas. I will try to contact support here in Aus

litters95 writes...

Hi Mams does not seem to work here in Aus for me? Any other ideas. I will try to contact support here in Aus

my mistake I do not not have CLI enabled... thought I did

Can you use C47H handsets from HK/China/UK with a C470IP bought in Australia?

HufflePuff writes...

Can you use C47H handsets from HK/China/UK with a C470IP bought in Australia?

Yes you can, I have just purchased a C47H from hip-ip UK.
Just the same except for the plugpack pinnout

Hmm HK or China, got a link to a supplier ?

Thanks zforum69 for your excellent posts.

I have been doing my head in for a couple of days trying to work out how to import contact details into the phone. None of the ways I have managed are either practical or elegant.

Wish I had found this post days ago. So far zforum69's method appears the best.

Since I have not had any measure of success I called Ingram Micro as they are now providing tech support for the Gigasets. I have asked them if they can provide me with a working example of a .vcf file with multiple entries so that I can emulate it.

My first impressions of this phone are very positive and everything apart from the directory importing works exactly as expected. I have had nothing but positive feedback about call quality as compared to the good old Draytek 2700VGL I have been previous using.

I will definitely be picking up some 675s while in the UK in July. Should eliminate the importing directory issues too.

Thanks again zforum69.

Atomique writes...

a working example of a .vcf file with multiple entries

Manually punch in a couple of phone contacts into your handset then from the web page settings/handset/save entries out and open the file with a text editor.

Make up extras with copy and paste, edit, then transfer that file back to your phone.

Bthree writes...

Manually punch in a couple of phone contacts into your handset then from the web page settings/handset/save entries out and open the file with a text editor.

Thank you Bthree!

Absolute pure simplicity. Functional and elegant! Genius.

Thank you very much!

DOH!

UPDATE – No joy even with the correct format. :(

As far as I can work out. 1 name = 1 number.

Hope that I am wrong or missing something simple.

Atomique writes...

UPDATE – No joy even with the correct format

Your save will create a file teledir.vcf

In this file you put entries like...

BEGIN:VCARD
VERSION:2.1
FN:TVSN
N:TVSN
TEL;HOME:132310
END:VCARD

BEGIN:VCARD
VERSION:2.1
FN:FRED
N:FRED
TEL;HOME:0212341234
END:VCARD

Go into the web interface settings/handsets

Browse to the file you just created

"Transfer" and the selected C47H handset will clock up the entries !

edit.... before you transfer its a good idea to do a "delete"

Thanks Bthree.

Your original response was a true piece of latteral thinking, I am kicking myself that I hadn't thought of it too.

However, my gripe with the importing the .vcf file is that each individual can only have 1 number assigned to them. Whereas the manual (page 128) indicates that each individual can have up to 3 numbers (general, office, mobile) plus additional info.

Considering these handsets have such limited memory, I don't want up to 3 separate records for one individual.

Your original response was awesome and now I am trying to get the phone working as per the manual.

Some great info here

I haven't read the whole thread as my questions are probably very easy to answer:

I currently use a Billion 7300G ADSL modem/router and I have my eyes on either the Siemens Gigaset S685IP or the new SL780.

Will the 7300G allow me to use this phone and VOIP services, or will I need to buy a VOIP enabled router such as the Draytek 2820vn?

Thanks

PVFC

pvfc1876 writes...

Will the 7300G allow me to use this phone and VOIP services, or will I need to buy a VOIP enabled router such as the Draytek 2820vn?

You don't need a VOIP router, just an IP/internet connection. The voip routers or other ATAs act as "gateways" to allow legacy analogue phones to use VOIP. The Siemens is genuine VOIP phone so simply needs to connect via a native IP/Internet connection.

Like all you other IP devices (PC, game console etc) you may have to adjust firewall rules and QoS parameters to get it working properly.

Z

Thanks mate

This is exactly what I needed to know

Cheers

Guys, Can I ask a really dumb question.
You obviously plug these into an Ethernet plug, do you just plug it into your router's IP port or do you need to plug it into the VOIP port of your VOIP enabled router?

Borizuka writes...

You obviously plug these into an Ethernet plug, do you just plug it into your router's IP port or do you need to plug it into the VOIP port of your VOIP enabled router?

just into a normal ethernet port. you don't require a VOIP enabled router for IP based phones. ie. either the VOIP modem/router will handle the voip functions, or the IP phone will. there is no need for both.

VOIP modem/router + normal phone
or
modem/router (without VOIP) + IP phone

= same result (the same functional result, many people report that an IP based phone has better call quality than a VOIP modem/router+normal phone.)

pvfc1876 writes...

I have my eyes on either the Siemens Gigaset S685IP or the new SL780.

for the SL780 you WILL require a VOIP capable modem/router, as it's not an IP enabled phone.

the current IP enabled Siemens phones are –
http://gigaset.com/shc/0,1935,hq_en_0_121551_rArNrNrNrN,00.html

Thanks again!

Hi Im thinking of getting the following phone Siemens C470IP Voip Cordless Phone for my naked ADSL+2 with voip connection which I will be getting very soon. At the moment I have a speedtouch 536 modem, I understand that all I have to do is plug the Siemens into this modem, but I want to use my present analogue phone as well what do I do? Is it best to get another voip enabled modem or just an ATA.
If I need another modem what would be a good recommendation?
Is the Siemens C470IP Voip Cordless Phone a good recommendation or is there something better?
If I have to get a voip enabled modem to be able to use my analogue phone does the Siemens just plug into that modem as well as the analogue phone?
I will be connecting with Internode, does the siemens have an answering machine added to the phone?
(ok I just noticed that nodephone has a free voicemail service anyway)

personally i wouldnt mix and match the c470ip with a normal analogue phone, as you lose many of the advantages of going the IP phone route.

you can get another gigaset handset (DECT/GAP compatible) that will connect to the c470ip base. which is the way i'll be going.

using an ATA/modem with VOIP AND the c470ip (while it probably would be possible) sounds like an invitation for difficulty. (ie SIP registration, QOS settings etc.)

Are there any news on whether the C47H handsets will make it to Australia?

FirestormX writes...

Are there any news on whether the C47H handsets will make it to Australia?

Time for a bulk order from the UK?

FirestormX writes...

Are there any news on whether the C47H handsets will make it to Australia?

The issue mainly was that the cost of the C47H handset was almost the same as the original unit and base station if you managed to get a discounted deal. The handset with the base station is a C47H.

vandozza writes...

personally i wouldnt mix and match the c470ip with a normal analogue phone

Thanks for the reply, I understand what you are saying and its becoming much clearer to me now. Thanks again.

So, are people still getting these from O/S?
Or do some local retailers sell the C470's or the C475's? If so can I get a price check?

EDIT: NVM, I am blind.
/forum-replies.cfm?t=1082794#r1

The C470IP base pairs, and works really well with an older A340 handset that I had purchased previously.

My c470 IP suddenly cannot see the BASE. it flashes "BASE 1" continiously.....tried re-registering the handset to the base and no luck....tried swtching base off and on again and that does not help either.....anyone??
Logged into base settings via webinterface and all seems ok – its online and checked software update > all up to date.....
what can i do??

Detail of my problem:

My handset is flashing "BASE 1" continuously and not connecting to the base.
I have tried the following:
a) Switched the handset off and on
b) switched the Base off and on
c) reset the handset
d) tried re registering the handset to the base
e) logged into the base from the PC and checked the firmware update section and it says firmware is upto date.
My current firmware is
Software

Firmware version: 021400000000 / 043.00 EEPROM version: 125

What can i do?

Mams see if you can find someone who owns a DECT phone and try registering your handset to it, or visa versa, to see if one bit has quit working.

I had this same problem with my C470IP and had to send it in to Siemens to get it repaired. The ended up sending me a new one. You can contact Siemens on 1300 780 878.

They could not tell me what was wrong with the phone.

mams writes...

Firmware version: 021400000000 / 043.00 EEPROM version: 125

That's interesting. That has been happening to me this morning as well, but have been able to just turn it off and on to get it to see the base again. I remember it happening to me before but was so very infrequent (once every 6months or so), so it was not an issue, however this morning it happened twice.

I upgraded the firmware last weekend. Does your experience correlate with a recent firmware upgrade (to 02140)? Anybody else who as upgraded their firmware to 02140 also noticed this?

Z

Hope i donot have to send the phone back as tht is gonna be a big costly, time consuming process....im sitting in Saudi Arabia and i purchased the phone in South Africa from a distributor....means getting it back to SA to the distributor who will send it to siemens.....

going to try finding someone with a gigaset that i can use to test which end is faulty.

Lastly, not sure what triggered the problem...i did a firmware update about 1 month ago(not sure how long exactly)
what actually happened here is that 2 days ago my landline and DSL both "died" and when i looked at the gigaset i saw this "base 1" flashing.
Now DSL and line are both up and running again and the gigaset has this problem.
Could it have done the firmware update on its own without me confirming it?

I have the Freshtel C470IP showing

Firmware version: 021230000000 / 043.00
EEPROM version: 121

The Data server is gigaset.siemens.com/chagall

Is the data server the same for non-Freshtel?

If not, you could try the Freshtel firmware?

If it breaks don't blame me!

Ian

mams writes...

Could it have done the firmware update on its own without me confirming it

Yes. I have an option 'Automatic check for software/profile updates:' that is checked.

Ian

data server is same....
i do have it on auto update so it seems that the new firmware is the culprit!

can i download an older firmware from somewhere??

Hi all, been following this thread closely as the C470IP looks like a great piece of kit.

Can someone please verify for me that it works just fine with MWI and voicemail from Maxotel? (Or failing that, another VSP?)

mams writes...

data server is same....
i do have it on auto update so it seems that the new firmware is the culprit!

I'm not so sure.

Firstly doesn't auto update still require you to confirm the upgrade?

Secondly the new firmware that you and I are both running on was released last month, so when you said you updated a month ago it may well have been this current one which also means you were running with it for a month or so.

Lastly my symptoms don't quite match yours. I can still recover with a power off/on. My issue may well be because I'm running on a low battery (I just moved so the charging cradle has not found it optimal location yet and thus the phone does not get charged frequently enough). I'll get it to full charge again and watch it a closely over the next couple of days.

can i download an older firmware from somewhere??

I've not tried this but it is documented in the source code. just over ride the user defined firmware file with the one you need. according to the gigaset website the previous version was 02139 so use the following url:

gigaset.siemens.com/chagall/2/chagall139_02.bin

Since you cannot connect to the base whatsoever it may well be your only option. Good luck.

Z

Does anyone know where the phone pulls its weather information?
Siemens report from Meteogroup.com, but the info on the screen never lines up with the BoM site – it seems to be closer to the Tulla Airport forecast rather than Melbourne Forecast, but its still not the same.
Ie Meteogroup reports forecast for Wed 12/19 deg
BoM Melbourne reports forecast for Wed 13/20 deg
BoM Tulla reports forecast for Wed 10/19 deg

I'm more interested in knowing if their data source is inaccurate/out-of-date rather than the last degree or two.

it might be 24 hours behind...

on an unrelated note, has anyone set one of these up with a draytek 1025w?
i changed my modem yesterday, and i'm having a terrible time getting any provider (node, mnf, pt) to STAY registered :(

I had a lot of intermittent problems loosing registration behind a billion with older firmware. I found a document on the Siemens site somewhere that suggested I set the 'Local Port' setting on the device to 49060 which I did.

Hasn't missed a beat since.

zforum69 writes...

I've not tried this but it is documented in the source code. just over ride the user defined firmware file with the one you need. according to the gigaset website the previous version was 02139 so use the following url:

gigaset.siemens.com/chagall/2/chagall139_02.bin

tried it by pasting that address in the user defined field on the web interface.....comes back with a reply that you cannot update as you already have the latest firmware.
THE FAQ on siemens website says that you can never install an older firmware...???
any ideas?

Peter Childs writes...

I had a lot of intermittent problems loosing registration behind a billion with older firmware. I found a document on the Siemens site somewhere that suggested I set the 'Local Port' setting on the device to 49060 which I did.

Thanks Peter,

To try this I guess I go to the C470's web config page, then to Telephony / Advanced, and change the SIP port from 5060 to 49060? (or/and do I have to change the NodePhone ports in Telephony / Connections / Edit / Advanced?)

Did you also change the RTP port from 5004 to something else?

Do I need to set up any port forwarding in the Dynalink 1025w? (I have turned it's SIP ALG OFF, as per one of your other posts.) I understand that port forwarding/QOS is probably unnecessary.

EDIT – also I'm trying to set up 3 providers (eventually), am I supposed to be using 3 separate SIP ports? ie provider1=5060, provider2=5061, provider3=5062?
Do I need to use separate RTP ports? (Im assuming not, but I'm not sure... All this was working well till the new modem was introduced.)

mams writes...

tried it by pasting that address in the user defined field on the web interface.....comes back with a reply that you cannot update as you already have the latest firmware.
THE FAQ on siemens website says that you can never install an older firmware...???
any ideas?

Sorry, I got the same message I even tried downloading it separately and then started a web server on my machine to see if getting the file locally would make any difference but no luck.

BTW, I don't seem to have those problems now that my handset is fully charged. Have you tried replacing the rechargeable batteries with a fresh set of alkaline batteries?. It wouldn't hurt to try considering there are no other suggestions out there.

Z

vandozza writes...

To try this I guess I go to the C470's web config page, then to Telephony / Advanced, and change the SIP port from 5060 to 49060? (or/and do I have to change the NodePhone ports in Telephony / Connections / Edit / Advanced?)

Just Telephony -> Advanced -> Listen Ports for VoIP Connections.

I found the source document
http://icm-km5.erlm.siemens.de/docs/cp/attachments/VoIP/nat-troubleshooting_en.pdf

Do I need to set up any port forwarding in the Dynalink 1025w? (I have turned it's SIP ALG OFF, as per one of your other posts.) I understand that port forwarding/QOS is probably unnecessary.

Although the Siemens documentation talks about port forwarding, I'm a bit fan of not doing it for Nodephone as we have equipment that will do hosted-NAT traversal very well without it (and its not just as simple as a single port forward, and depending on your router/NAT you can introduce another set of issues with media ports)

EDIT – also I'm trying to set up 3 providers (eventually), am I supposed to be using 3 separate SIP ports? ie provider1=5060, provider2=5061, provider3=5062?
Do I need to use separate RTP ports?

I don't believe so. I, however, only use a single provider so my opinion might not be worth much in this respect ;)

All this was working well till the new modem was introduced.

Rollback?

Thanks for all the suggestions Peter.

I now have things up and registered, and they look to have stay registered for 7 hours.

However it seems to be a combination of things, changing the port to 49060, and removing the DHCP reservation in the router.

If I now set the C470IP a static IP (either from making a DHCP reservation in the 1025w, or by setting this in the C470IP's config pages) it results in services that WILL NOT stay registered. (and the C470IP won't appear in the modem's DHCP table.)

Moving the C470IP back to a dynamic IP fixes this issue immediately.

I was assuming I would require a static IP for port forwarding / QOS. If this is not necessary, I'll just leave the IP as dynamic, and have a working service :)

I can't figure out why the C470IP doesn't like a static IP though! (am I meant to be setting the static IP both in the modem and the phone? or just on either side?)

anyone with a solution for my C470 ip that is still flashing "BASE 1"?

did you reset the base back to factory default?

- pull the power plug on the base
- hold down the blue paging key on the base
- plug back in the power cord (while still holding the key)
- hold key for 5-10 seconds

you then have re-pair the phone.
- put handset into register mode
- hold down blue paging key for another 5 seconds

from – http://gigaset.com/shc/0,1935,hq_en_0_142886_rArNrNrNrN_PRUNIT%253AIQ-ul-GC470IP_PRODGRP%253AHO,00.html

vandozza writes...

I can't figure out why the C470IP doesn't like a static IP though

I have run my C470ip with a static ip from day one without a problem

1) easy to find
2) Billion use ip address for QOS, not that I find QOS on Billion 100% effective

BTW I use standard port settings, out of the box

sorry my post shoud have read "I don't know why the C470IP doesn't like a static IP with a Draytek 1025w modem"

All was fine using an old dlink-300+ bridged to a dlink dgl-4300 router with a static IP.

UPDATE on my issues – was using firefox to configure C470IP web pages. Seemed really slow/clunky.
On the off chance i tried Safari, and I seem to be having more success. So maybe my issues have been more FF related than phone/modem/IP...?

Here is a note that I'm going to hide in this thread on the off case that someone will find it when looking :)

After getting a fresh new c470 I initiated the firmware upgrade. A long time later I noticed it still was flashing away... hmm....

I threw a sniffer on the link and noticed that it would stop and start downloading firmware, and never seemed to complete the upgrade. After looking a bit closer I noticed that if the device performed a DHCP renewal whilst upgrading the firmware it would stop the firmware upgrade (and start it again).

I adjusted the locate DHCP server (it had a 10 minute renewal time.. yes a bit silly) and on DHCP ACK it just stopped dead. After a power cycle it started the firmware upgrade again and then walla ... all fine.

My handset somehow had been de-registered from the base so I just re-paired it and off and working again.

Just an update on my experience as well...
I've now been successfully registered for almost 24 hours, where previously I couldn't get past 5 min.

I had noticed when using FireFox 3 to edit the C470IP web config that things were running pretty poorly, slow page loads, sudden CPU usage spikes, general internet browsing laggy.

On the off chance it was a FF problem, I tried Safari.

I now have the C470IP assigned a static address (an IP reservation in the Dynalink 1025w, with the C470IP still set to "Obtain Automatically")

I think for now my huge issue of getting a constant registration has been solved.

Moral of the story – If you are having trouble with the web config, try something other than FireFox 3.

vandozza writes...

did you reset the base back to factory default?

tried the process you explained....still no response...
also tried registering another handset AS14 to the base and no response....

and you are sure you are completing the registration correctly?

- go to settings / handset / register on the handset.
- enter pin
- hold down blue paging key for 5-10 seconds
- accept the registration on the handset.

if you are doing that correctly, the I have no idea what the problem might be :(

I have a Freshtel firmware F470IP.

I set a dial-plan to allow me to dial 771 (Pennytel balance enquiry) so that the local code should not be inserted but it is. So I see 03771 has been dialled and get a message about Optus not recognising the number.

The dial-plans work with numbers beginning with '0' but don't work with service numbers such as 771.

Is anybody else seeing this?

Interestingly, network mailbox numbers work OK and don't insert 03.

Ian

have you had a look at the setting in the web config pages, there is an option on the dial plans settings to insert area code automatically.

you didn't turn this on?

sorry i can't be more clear, but I'm not at home near my unit.

vandozza writes...

and you are sure you are completing the registration correctly?

doing exactly as per your instructions....
when i hold down the blue paging key on the base, nothing else happens....

How do i know that the base is going into registration mode? the blue light does not flash...it si just a solid blue light if the LAN is connected and if not connected, it is not lit up at all.

Siemens not even replying to my query....
wonder if i should wait till new firmware is released or if i should send the phone back to South Africa

mams writes...

How do i know that the base is going into registration mode? the blue light does not flash...it si just a solid blue light if the LAN is connected and if not connected, it is not lit up at all.

yeah it just stays blue afaik. the only feedback that you really get is that the phone reports that it is ready to register.

do they release firmware often? i've only had my unit 1 week.

The same issue "Base 1" flashing occured on my C470 in late december (2008) so i think it was running firmware 021390. I had it set to automatically check for firmware/profile updates.

If this issue is occuring for users of the new firmware, the issue might exists with the latest firmware as well.

zforum69 writes...

I've not tried this but it is documented in the source code. just over ride the user defined firmware file with the one you need. according to the gigaset website the previous version was 02139 so use the following url:

gigaset.siemens.com/chagall/2/chagall139_02.bin

I couldn't resist and just upgraded my Freshtel firmware to this one. The updater is now informing me I have the latest firmware which isn't correct as some of you have one higher.

I did it because I had an issue with dialling plans (which I still have) and I notice the Freshtel phone does not give access to the Gigaset services (which it still doesn't even with this firmware).

I have asked Siemens Gigaset what to do.

Ian

vandozza writes...

have you had a look at the setting in the web config pages, there is an option on the dial plans settings to insert area code automatically.

Yes, I have this turned on.

The manual says a dialling plan does not insert the local area code but mine is when I try to dial 771.

I have flagged it to Siemens and will wait to see what they say.

Ian

Iangh writes...

Yes, I have this turned on.

does it work properly with this option turned off?

that would pretty much indicate that it IS a firmware/software bug.

(I have mine set to off, as my providers NodePhone and PennyTel seem to do this automagically.)

vandozza writes...

does it work properly with this option turned off?

Yes for Pennytel.

No for MNF. Unlike PT you need to set inserting the local code.

I think it's a firmware bug as the dialling plans work when I use a number beginning '0'.

I got this phone based on the recommendations here, and have to say that I am very happy with it. I'd like to get some extra handsets though, so if anyone has any advice on how to connect other brand's handsets to the base station of the C470-IP, would you mind sharing it?

I have an older 2.4GHz Uniden cordless which probably won't work with it, but I know that some Doro handsets do. I was just wondering how people go about registering the handsets with the base station. Thanks in advance...

hi,

does anyone know how to increase the "mic" volume so that parties can hear you louder?

thanks

Rudi writes...

I was just wondering how people go about registering the handsets with the base station

On the handset get it to go into registration mode (this of course differs depending on the make/model of the handset). On the Siemens base stations press and hold the one button for about 3 seconds.

If it works the handset should tell you it has registered. You can then go into the web interface and give the handset a specific name and associate it with a provider(s) etc..

OK, so it really is a straightforward as that. Good to know, now all I have to do is decide on which handset. I noticed that there are some really cheap Siemens Gigaset handsets out there, might try one of those...

In case people don't know, Siemens has sold most of the Gigaset business to ARQUES as of Oct 2008. The new company is called Gigaset Communications, so good luck having Siemens replying your support calls.

http://de.biz.yahoo.com/01082008/356/dgap-news-arques-to-acquire-80-2-of-siemens-home-0.html

theguru1 writes...

The issue mainly was that the cost of the C47H handset was almost the same as the original unit and base station if you managed to get a discounted deal. The handset with the base station is a C47H.

i have my C470IP sitting at home and considering getting this before I even unpack the box

http://www.ryda.com.au/Siemens-C47H-Additional-Phone-Handset-p/c47h.htm

will make the system far more user friendly for us!

brentonsav writes...

i have my C470IP sitting at home and considering getting this before I even unpack the box

http://www.ryda.com.au/Siemens-C47H-Additional-Phone-Handset-p/c47h.htm

will make the system far more user friendly for us!

Considering nobody has been able to get them from Australia to date I had to be a little skeptical. I could not help myself and rang them and they said they have no stock and anticipate some at the end of the month.

Maybe Australia is finally going to get some supply of these additional handsets?

After using my set for a good year now I definitely would get additional C47H handsets. There are several things that can't be done with a generic DECT GAP phone as they use the handset's soft keys like being able to transfer between between handsets, 3 -way conferencing, etc. We also use the handset for the weather.

Each individual need to to make their own assessment whether it is worth it. I figured ff you are willing to pay the extra for C470IP instead of an ATA with analogue cordless phone, I suspect you would also be interested in paying extra for the additional handset.

Z

zforum69 writes...

Each individual need to to make their own assessment whether it is worth it. I figured ff you are willing to pay the extra for C470IP instead of an ATA with analogue cordless phone, I suspect you would also be interested in paying extra for the additional handset.

Bloody oath. It's the way of the world.

I'm about to connect naked ADSL and using VoIP and love the thought of not having a Tel$tra bill at all!

i still have a BRICKED phone.....anyone out there with a solution?

mams writes...

still have a BRICKED phone.....anyone out there with a solution

Yeh, sell it to me.... Ha

.... only chance is to send it back to Siemens

zforum69 writes...

I suspect you would also be interested in paying extra for the additional handset.

You may wish to see this post: /forum-replies.cfm?t=1033963&p=17#r327

it may actually be cheaper to sell it then send it backl to SA and get it back to me again...lol

Careless writes...

The new company is called Gigaset Communications, so good luck having Siemens replying your support calls.

My C470IP base station has apparently died on me. The router ENET port interface light is off...not a good sign. Powered the base on and off a few times, tried different electrical sockets. No multimeter to check the adapter is not at fault, but wouldn't expect it. The C47H handset seems to be working fine, cannot detect any base signal.

The Nokia Siemens Networks number included in the box is not the correct one. JB HIFI tracked down the right number, identical to that reported by PK above (a different PK...): 1300 780 878.

They said to post back the complete contents to them in Burwood VIC, at their cost (reply paid). You need to get a reference number from them and put it on the package label. So for the time being, Siemens are still providing support for these phones. They did ask when I bought it, but I assumed that was just to check that it was still under warranty.

I also asked about additional C47H handsets...they said they had a batch of them on order, and they'd be sold through retailers in due course.

Have those interested in the c470 seen this Freshtel deal atm:
/forum-replies.cfm?t=1164042

Good deal – this would suggest the product is also being discontinued in Aus as well as UK?

I hope they give you a version of the A580IP to replace if so!

JP

My Siemen C470ip with two handsets changed back to EST at 11:30 this morning

Only a week early ! well in Vic anyway

Along with my Telstra mobile, did the same, strange !!!

I'm glad somebody else struck the same problem. Got my two handsets on Friday, every OK on Saturday. Then Sunday came and I've been trying to figure out the incorrect time display, new firmware, time servers and what not. I've now updated to the the latest (02140000000/041.10 Eprom version 125.) but the problem is still there.
By the way, how do you force a time synchronisation? Pulling out the Lan lead?

sardine writes...

By the way, how do you force a time synchronisation? Pulling out the Lan lead?

Just change the time setting and save. However, it wont give you the correct time. It will still be an hour behind

Recent review on C470IP
http://forums.mactalk.com.au/18/61940-seimens-gigaset-c470ip-voip-cordless-phone-review.html

I have a DLink Dir 635 router and a Siemens CP470IP. If the system looses power, the only way i can get the CP470IP to re register is to turn off the router, and CP470, disconnect the CP470IP from the router, turn on the router, wait until it is online, then connect the CP470 back to the router, and lastly reconnect the power. Finally it reconnects and registers.

I also have an old Grandstream Voip and it does the same thing.

Is there a setting in the router or phones that needs changing

Any help would be gratefully accepted

Thankyou in advance

Alex

Alex111 writes...

s there a setting in the router or phones that needs changing

Is the Ip set to dynamic?

Tried just disconnecting the base station Ethernet connection?

Alex111 writes...

If the system looses power

Which system, the phone or the router?

My only guess from the information you gave is to check for any IP address conficts.

The IP setting was one thing I was wondering about. At present it is set to dynamic. I didnt know if having a static address would be more stable.

It does the same thing if the internet goes down for a short time.

Yes, Ive tried all combinations of reconnecting, but nothing works until the router is powered down and everthing reconnected as mentioned.

The Grandstream isnt so bad. If the internet goes down, it will reconnect ok. But if the power goes off, I have to follow the above mentioned proceedure.

Finally got it going with help from my Voip provider, Alex at MaxO.

We stopped the router being a router by plugging in the Modem into one of the spare ports on the router. Not the internet port.

This made the modem assign IP addresses to everything and instead of the router reassigning IP addresses to the phones

So far, so good.

Alex

I'm having an adventure with dial plans.

I can add a dial plan to send "000" calls through my land line without any problems.

If I then try to put in a dial plan to specifically direct a call to a mobile number (i.e. 04nnnnnnnn) through my land line..... after hitting the Add button, the C470 ISP web page for setting dial plan stops displaying any dial plan information at all (and some other aspects of the page, such as the country set up information at the top are also no longer alterable).

I have to do a reset on the base station to get the Dial Plan section to display again.

I'm using Firefox 3 ro view the page and the base station is running the current firmware release.

Any clues as to what is happening would be greatly appreciated.

I have had a C470 for about 6 months and it is working fine on MNF and other vsps.

I have recently changed to Internode and was going to use nodephone with it.

The problem:
I have the c470ip registered to nodephone in the web configurator but when I dial a number it starts to dial and then flashes: No Ip connection

I know I have an IP connection as if I un-tick nodephone and tick MNF, MNF then registers I can make calls on MNF.

I have contacted nodephone and they can't help.

Billion 7404 VGPX Router.

Also tried it on my old Netgear WGR 614 which is still on Optus cable and the same thing happens.

It seems to me that it must be something to do with the Nodephone setup.

Has anyone any ideas?

petegus writes...

I have contacted nodephone and they can't help.

Internode sell them so they must be able to support them

http://www.internode.on.net/residential/nodephone/hardware/

Try contacting them again

petegus writes...

Has anyone any ideas?

See.... www.internode.on.net/sup.../siemens_gigaset

Thanks Software tools and Bthree.

I have loaded the nodephone profile from the gigaset and also printed out the Siemens gigaset info.

I also talked to the Internode nodephone help desk yesterday afternoon for about 40 mins but they have not seen this problem before.

They are going to ask the nodephone second level people on Monday if they have any suggestions.

petegus writes...

Has anyone any ideas?

Have you disabled the SIP ALG (Under firewall option in the Billion)? Firmware on the Billion is?

Have you set the 'Listen ports for VoIP connections' SIP setting to 49060 on the Siemens under Settings -> Telephony -> Advanced Settings.

Thanks Peter,

Yes, I checked the latest firmware on the Billion site – 5.53 S5 is installed.
It is brand new – made in 03/09.

I have disabled the SIP ALG – it made no difference.

I saw that Siemens paper on changing the ports to 49060 but as soon as I change any one of them I get: Server Not accessible

I also re-loaded the nodephone profile on the Siemens.

I also disabled both the Windows Firewall and the Billion firewall temporarily to see if it was a firewall problem but that also made no difference.

It seems to me you have isolated it to a nodephone connection problem. Just to make sure can you make x-lite work with nodephone?

Z

petegus writes...

I saw that Siemens paper on changing the ports to 49060 but as soon as I change any one of them I get: Server Not accessible

I think you are changing the ports on the wrong page. I did this the first time.

You need to change it on the following page:
Telephony -> Advanced Settings
SIP port and RTP port.

Thanks Guys,

Still no luck with the Siemens.

Yes Yert,
I tried the advanced settings page as well.

I thought I would try another method and so I used the 7404VGPX and can log in and use Nodephone with an old Telstra handset on Port1.

At least I have proved that it is definitely the Siemens setup at fault as it works on the 7404VGPX.

petegus writes...

and the Billion firewall

The Billion firewall is quiet restrictive. From what I have seen you need some allow entries for both the sip and rtp traffic with the Siemens.

I had something like this (I'm going off memory...)
SIP – source 192.168.0.1/255.255.255.0 port 49060-49060 dest 203.2.134.1/255.255.255.255 port 5060-5060
RTP – source 192.168.0.1/255.255.255.0 port 5000-5040 dest 203.2.134.1/255.255.255.255 port 32000-63000

Checking the Billion f/w logs helps. Also the Siemens 'rings home' to check for firmware updates so you might want to allow it http access (depending on your fw setup).

Personally I would disable Billion fw, turn off SIP ALG on the Billion, set the local port to 49060 and then give it a few minutes to settle, then check out how things look.

Well some success.

I did a hard re-boot of the Siemens and re-entered the setup. It will now ring out.

I only entered Nodephone.

I have the SIP port as 49060 and the RTP port as 49004.

It will not allow a call to my nodephone number – either on the Siemens or Port 1 on the Billion – it has an engaged tone all the time when you call it.

Fw is Disable and the SIP ALG is Disable.

petegus writes...

It will not allow a call to my nodephone number – either on the Siemens or Port 1 on the Billion – it has an engaged tone all the time when you call it.

Is it a NodePhone1 or NodePhone2 service? NodePhone1 service's do not accept PSTN inbound calls....

Given both your Billion and Siemens units are Internode recommended products I would recommend trying our support guys again to get them to run some tests on your failed calls.

Re the "No IP Connection" problem.

I have two VSPs, Nodephone and Maxo. I consistently get the "No IP Connection" error when dialing out on Maxotel. Nodephone outbound is fine. Inbound is fine on Maxo.

I've contacted Alex at Maxo but he didn't seem to know what the problem was.

I'm using a linux (OpenWRT) firewall, but no special config for VoIP (other than QoS).

Hi Peter,

Yes I talked to support this evening and found out that if you don't have Nodephone2 you do not have inbound PSTN calls.

I have re-installed Nodephone as #1, Mynetfone as #2 and Sipme as #3 on the connections page.

They are all registered simultaneously.

I can make a call on any of them by using the override eg 0294131234#2 for MNF.

I have also changed the SIP port back to 5060 and the RTP to 5040 and it works like a dream.

So what caused it? I don't know but my suggestion is that if anyone is having a problem with the Gigaset is to do a hard reset and start again.

Very frustrating!

Thanks again for your help.

Hi Guyz,

i just got this phone and wondering how do we make calls to other users connected on gigaset network, i tried making a random call and i get IP status code x194 error or something like that. normal outgoing calls through penneytel are fine.

any ideaz ?

thanks

hiq writes...

i just got this phone and wondering how do we make calls to other users connected on gigaset network,

You just enable it in connections configuration and you dial another gigaset nubmer with the suffix #9. The number is the same ans the number you sue to login into the gigaset portal to configure things like weather. It is in the "services" part of the C470IP web confugrator.

It is not that useful to us here as the latency to make a call (going from Australia to Europe and back), would make the call unbearable, this will happen with every call to a gigaset number even if both of you are here in Australia. If you have a somebody you want to call in Europe who also has a siemens IP gigaset phone, then that would be a different mater.

Z

Hi I was thinking of getting the S685Ip I already have a 470IP I guess there would be no problems registering this to the 685 base sation with all features?
thanks

litters95 writes...

Hi I was thinking of getting the S685Ip I already have a 470IP I guess there would be no problems registering this to the 685 base sation with all features?

thanks

shouldn't be a problem to register. I was setting up my s685ip last night and was pleased to see that you can register any of s45, sl37h, c47h (which should be your handset?), c67h and s68h. Not sure about whether it would adpot the 685 features though? you might be able to find the answer at the
gigaset website

FullTimeMum writes...

shouldn't be a problem to register

Thanks I tought it should be Ok but just wanted to check. Roughly what do they cost delivered to Aus. I sent an email to Hip-Ip but havent heard back yet

zforum69 writes...

It is not that useful to us here as the latency to make a call (going from Australia to Europe and back), would make the call unbearable,

I have done a few calls using the Gigaset network from work to home and the quality was fine.

theguru1 writes...

I have done a few calls using the Gigaset network from work to home and the quality was fine.

That's surprising, I notice the latency with voip calls to a mobile network, so I assumed the gigaset network would not be good enough to use often considering how cheap it is to call using a VSP anyway. Based on your experience I might just have to give it a go if I one of my family or friends gets a siemens gigaset IP phone.

Z

zforum69 writes...

might just have to give it a go if I one of my family or friends gets a siemens gigaset IP phone.

Z

Happy for you to give me a call to see what the quality is like. Whim me.

litters95 writes...

Roughly what do they cost delivered to Aus. I sent an email to Hip-Ip but havent heard back yet

you should be able to get a qupte going through their checkout process before actually completing it.

Anyway, GBP12.99. FYI, there is a thread on the s685ip here

FullTimeMum writes...