DIDLogic will use a CLI that you have through them, but they make you jump through hoops to use a geographic number from elsewhere. Am surprised they will use your mobile number - they don't offer anything on the menu to me other than numbers they provided.
iOS has a file system, but I've never needed to access it.
Choice of codec is important - G711 is often the default won't run well over 3G, though gives great quality on faster networks. Try GSM or iLBC. Make sure you disable all the others otherwise a negotiation might occur that means you're not using the codec you think you are, though the better clients will tell you what they're using. Make sure your sip trunk supports whatever you try too. But yes, latency can be a big issue - a (low bandwidth) ISDN line has low latency.
I got the mobile # CLI without asking for it. It was a default setting of some sort, IIRC. Under SIP I see a pulldown menu for CLI and in there is
No CLI
My mobile #
My geographical number
Sure the Iphone has a file system (it runs some version of BSD unix?). I had a good nose around after I did a jailbreak on Justine's Iphone. Didn't do me any good because most of it was obscured by the use of bland all-numeric directory names. But it's clearly Apple's policy to not present it to the outside, from within any app. It works for them and their users; you have "Camera Roll" instead of "c:\DCIM\100NOKIA" 
Fortunately they leave a door open for PC apps like Iphone Explorer which, even though it has access limited to 3rd party apps and only to some directories like Documents, can be used for some "extremely interesting things" 
I will try the other codecs and disable the higher-perf ones. I have found the ones I use sometimes fail to work even on what is supposed to be WIFI.
I am still trying to find out from them whether the SIP session can be kept open just by sending a port 5060 packet to their IP (which would be fairly easy to do) but they are avoiding replying to that direct question.
I will try the other codecs and disable the higher-perf ones.
G729 is the best for low bandwidth connections like 3G but because most implementations require a licence fee you mind not have it available. If it's available you usually have to pay a couple of pounds to use it, but I'd say it's worth it. Otherwise stick to GSM, though the quality is pretty low it's reliable.
I am still trying to find out from them whether the SIP session can be kept open just by sending a port 5060 packet to their IP (which would be fairly easy to do) but they are avoiding replying to that direct question.
I don't think DIDLogic will help you with regard to sip sessions etc.
You don't actually maintain a TCP/IP "session" or connection when using SIP (i.e. when you're not actually in a call). SIP works like HTTP - it's request and response.
When a call comes through to your SIP provider they need to be able to reach your SIP device to tell it that there is a call. You can either tell them in advance where to find you (e.g. a static IP address) or you maintain a SIP Registration.
Once they know where you are, they know where to send the call. However, for it all to work, two things need to be considered. Firstly they need to be able to get reach the device in terms of firewall penetration. At home you can forward the ports on your firewall to the device. Or if you've initiated a connection by doing a SIP registration then they will (simplistically) "come back to you down the same pipe you went to them on". But for that to work that connection must still be open. It's no good just sending a packet from some other device on your network to them, or even from the phone to them - it's the actual socket that you opened originally that has to stay open. Some SIP clients will do that. When you're out and about it must use registration (because of IP address changes), but at least then you have no firewall to worry about.
Secondly, your SIP client has to be listening. A VoIP phone or a soft client on a computer will be, but on a mobile phone it depends on the phone OS and the phone client. On iOS they can't listen for that when in the background, hence the use of Apple's Push Notification Service. I don't know about the Nokia SIP client.
So really it all comes down to your SIP client. If you want to try your SIP client against our VoIP server so that we can see what packets are actually being delivered and when, let me know.
As for DIDLogic and their CLI, the only options on my drop down list are the geographic numbers I have from them. They won't port in existing numbers from elsewhere because they're not a phone company (so they say). Shame really, their quality seems fine and they are cheap.
I can't disable codecs but can change their priority, and currently have G729 at the top.
The others are (in order) PCMU PCMA AMR-WB AMR iLBC CN.
I can get incoming calls to work but it doesn't work for very long because
The first I can deal with by port forwarding 5060 to an internal IP which is the IP of the phone, and that works. OK, the phone gets a dynamic IP over DHCP but I can do this to prove the point. If I was only ever doing this at home I could give the phone a fixed IP on WIFI. To make it even more secure I could tie the incoming traffic to come only from DID's IP, etc. otherwise every sniffer will be banging my router on 5060.
The second needs some kind of a keep-alive process running.
I wonder how I managed to present my mobile number CLI then Without that, the whole thing would be useless.
G729 at the top is a good plan, but I don't know if DIDLogic support it. If they don't you'll be using PCMU which is a bad choice over 3G. Put iLBC second from the top.
Can you tell your SIP client to register? And set the registration timeout (I've never seen a SIP device where you can't set the registration timeout)? This should solve the second problem.
I haven't told DIDLogic my mobile number so maybe that's why they didn't put it on the available list. I don't use them for anything important so I'm not that bothered, was just curious as to how your number ended up as an option really.
They say they support G729:
On 3G, you might wish to change codec from G711 to G729, will fit into the
100 kbit/sec stream perhaps.
On HSPA, generally any codec will work, but sometimes G711 has issues over
3G. On WiFi, only use G711 - best voice quality.
Re the CLIs, this is what I have
You're going to create several SIP logins and each numeric login will be assigned a CLI you wish to send. Use the numeric logins on separate devices and send the desired CLIs from each.
I do have a reg timeout, which is set to 1800 seconds, which is curiously the same as the DID login timeout, but I suspect this is a coincidence since setting it to shorter times doesn't seem to do anything. There is a huge amount of stuff on google about this; the usual stuff where people are tearing their hair out because nobody is responding
Have sent you an email with details of connecting to our VoIP server.
Just a little update, for those who care
This VOIP stuff turns out to be hugely complicated, if you want incoming calls to work and it's a mobile client device and you want to make it work over both wifi and 3G.
It's clear that most people that use VOIP and who get it working, do so due to
They are in a corporate scenario where somebody worked it out and deployed a config via a download, or
They use a client app supplied by the VOIP provider, which comes preconfigured
They use a common client app that uses a commonly used service and "just works" - Skype is perhaps one example
They make outgoing calls only (the config is then relatively totally trivial)
They are not doing VOIP on a mobile client (are using a PBX)
There are two keep-alive process timers which need to be working. One is a frequent one - of the order of 30-60 seconds - which stops the router closing the NAT channel. Most routers will close it in 60-180 seconds (mine is 180) and it's usually not configurable. The other keeps alive the login with the SIP server, which in my case gets closed after a time specified by a config parameter in the phone, which I can set to a maximum of 3600 seconds.
I eventually found the first one. It appears under "NAT firewall" settings. I set that to 55 secs, and it clearly works, and works for ever. It keeps the NAT hole open long after the SIP server has disconnected you However it works only with WIFI.
It doesn't work on 3G, which (Vodafone) kills the session after a few minutes. I came across some snippet from Nokia suggesting they test for the phone being on a LAN subnet which presumably means having a client IP of 192.168.something (how else could they tell??) which is a dodgy practice and would prevent it working on 3G where the client IP will be whatever the cellular network gives you. This is not an issue since I am unlikely to want to use VOIP over 3G (why would one, unless one's call allowance has run out but you still have some data left). One can understand Nokia having done this, years ago, because historically the mobile business would have gone berserk at the thought of VOIP over 3G. OTOH the reason could be totally different...
The other keep-alive process is probably in there somewhere but I haven't found it yet. It may be something implicit deep down in the Presence settings, or something called XDM, or something called VCC. If one googles on this stuff one just finds a load of people tearing their hair out.
DIDlogic has stopped providing support. I think they have a quota of email replies But I like them because they are very cheap, they are PAYG (so you can't get screwed), and they work.
SOLVED.
One has to config the STUN server as well, with an IP of 0.0.0.0 and NO other config.
Obvious, really
It works on 3G too.
BT have a free Android app that lets a landline, phone and broadband customer (me) make calls over wifi (home or hotspot) on my domestic tarif, from anywhere. Just tried it - it works.
