sip client os2008
 

I am curious to know the experience of other people who are using the built-in SIP app in os2008.

I can make call from the app with no problem, but having problem on incomming calls.


I know there is no issue with my router firewall as i m using the ATA device behind the same router.

is SIP app in os2008 keep the connection with the sip provider or only make connection on outgoing calls?

Re: sip client os2008
 

The OS2008 SIP client will register on the SIP server/provider and will attempt to keep the connection alive. I'm using it for both incoming and outgoing calls via an Asterisk server.

Chances are this will have something to do with your SIP server configuration. What are you running for a server? Are inbound calls routed to this SIP account when your using a client other then the Nokia?

Re: sip client os2008
 

well, i have two a/c from the same sip provide. my home phone from ATA I am calling to my other a/c in nokia, no response.


If I call from nokia to my home phone, it rings.

Have no clue what I am doing wrong,

what is the value you have: "keep-alive mechanism"?
do you have any frequency defined for "keep-alive frequency"?

Re: sip client os2008
 

I'm using the default settings for the OS2008 SIP client for keep-alive, lookup public IP, etc.

I'm not clear on what you mean by "a/c". Accounts?

What SIP server/service are you using?

Is it a server at home behind your firewall/NAT or one out on the net?

Re: sip client os2008
 

yes a/c = account

I m using gizmoproject sip account.

I have no problem making outgoing calls, but no incomming on my nokia.

My ATA is fine and receiving all calls on the same network and same settings.

Re: sip client os2008
 

I am able to "register" with my SIP provider (Free, french ISP), but no way to get outgoing/incoming call working. As far as I know it is a known issue, but not solved yet (https://bugs.maemo.org/show_bug.cgi?id=1699)

Re: sip client os2008
 

I've also got mine registering to an asterisk box and I'm having no trouble with incoming or outgoing. I've used the default for keep-alive, didn't change anything. And, much to my delight, DTMF now works!

It sounds like since you have two accounts, maybe thats the problem with your Nokia device. Most firewalls do NAT, and if you've got two different machines on the network trying to talk SIP, that might be the issue.

Re: sip client os2008
 

well thats not true, as I am using actually 3 ATAs inside my home network, and there is no problem with any of them (I know sounds weired but its from 2 different provider who doesn't offer the SIP forwarding).

So the only problem is with the N800. I have to check that in my nokia as well as I have 2 accounts setup in the sip client (only 1 is enable). I guess I remove that disable account and then will check it again.

But this problem is something only with the application setting and not with the network, I am 100% sure on that.

will post the result once I will check that.

Thanks...

Re: sip client os2008
 

I think I know what it is. If you have gizmoproject and have inactive a/c you don't get ring (no incomming).

Thats my guess have no way to check that though.

Any idea how to activate the account without buying anything from gizmoproject?

thanks once again.

Re: sip client os2008
 

Thanks for this post. I've been wondering/thinking the same thing. Gizmo has this 'all calls free' activation process that involves inviting others and using the product, etc, and I've done both of those, but somehow one or the other account gets labeled 'inactive' and then no ring. I also have not been able to verify this, but it looks like they want us to buy something in order to make it work.

The other thought is that the no ring problem on Gizmo has something to do with router settings, and this is way outta my league. I have a simple little wireless router (netgear) at home, but I've never tinkered with the settings. Any other ideas?

Re: sip client os2008
 

well, as I mentioned above that I do have two accounts from gizmoproject. one is paid which is obviously active status and another one is just a dummy account which I was planning to use it from my nokia to have atleast incoming calls.

in my router I just open couple of ports:
GZ RTCP : Port 5005
GZ SIP : Port 64064
GZ RTP: Port 5004
GZ SRS: Port 7070
GZ_SIP2: Port 5060

and the accounts is registered with my ATA in the house, and have no firewall problem at all.

I was fighting with this N800 thing and using the in-active gizmoproject account which was not ringing at all, and when Imake use the active account, it rings, so the problem is the ACF not the N800 SIP client.

anyway, i tried all the ways to the status active, i.e. inviting people, adding in my contact list etc, everything except to buy the credits and nothing bring my account to the active status.

HTH.

Re: sip client os2008
 

Thanks, Shyboy for the info. I'll look at my router settings at some point, but I'm still thinking it has something to do with the Gizmo status. Just like you I have the two accounts. Interestingly, I was able to connect them a couple of times, tho, while here at work, but never at home, which is why I'm still considering the router question.

Re: sip client os2008
 

Not sure if this will help you or not, but try enabling port forwarding in your firewall router (the Netgear should have a screen for port forwarding) Find it and forward UDP port 5060 to your N800, or other SIP device.

For example, I forward 5060 from my ITSP (viatalk) to my Asterisk server on my local network.

Re: sip client os2008
 

Thanks, Mitch. I appreciate the help!

The first part is something I can try, but I am not so sure about the last bit;). My 'system' is pretty basic. I have a cable modem (Time Warner) and a PC with a wireless adapter.

Re: sip client os2008
 

well not really a port forwarding, because if the N800 is taking IP through DHCP then the port forwarding will require the ip address to forward the port to.

check and see if you have any port triggering tab in you router setup.

if not that take the suggession what deeteroderdas suggest and
in addition to that, you need to assign the static ip on your N800 advance settings in your connection manager, in that way you will always have the same ip from your router where the port will be forward.

IMHO this is not a good idea. :)

Re: sip client os2008
 

Agreed.:)

What you could do, and what I do, is let the N800 stay DHCP, then go in the router and tell it to always give the same IP address to the N800. This is done based on the MAC address.

My router is an ASUS, running Linux/OpenWRT but the Netgear should have a similar capability. I used to have a Netgear (631?) and it had that ability.

Re: sip client os2008
 

but remember that, according to this way, only N800 will be capable receiving the SIP calls, if later you added the ATA device or any other computer running the GizmoProject Software, it will not be able to receive calls, as you can only do the port forwarding to one IP address.

I would suggest you to buy some smarter router (I got Linksys WRT54GS, with third party firmware) and use the port triggering not port forwarding.

BALL is in your court now.. :)

Re: sip client os2008
 

OK, I'll punt...:)

I actually have my N800 registering with my personal Asterisk server, which is running in a VMWare server instance on my Linux box. Additionally, my Linksys ATA the antique analog phones are registered to Asterisk, as is the Polycom SIP phone. The Asterisk server registers to my ITSP and is means for making calls in and out.

So, port forwarding works for me...YMMV.:D

Re: sip client os2008
 

well, yeah in your case its fine as you are running the asterisk server, so you forward those ports to ashterisk, but I made the valid point though :-)

Re: sip client os2008
 

Here is my current experience but my friends had working setups at home:

http://internettablettalk.com/forums...ad.php?t=10344

I really can't figure out what is wrong when ekiga and 2 ATAs work flawlessly but the n800/2008beta doesn't and is quite erratic.

Re: sip client os2008
 

Hi.
I am setting up N800 instead one of two computers. Computers are running soft phone sip clients (X-LITE). Accounts serviced by outside VOIP providers. I can call from one account to another (2 computers setup scenario) by using PSTN dialing 1(XXX)XXX-XXXX and VOIP dialing XXXXXXXXXX@PROVIDER.COM/NET. After setting up N800 replacing one of computers I am receiving calls to N800 and can dial PSTN number to my second account. However attempting to dial VOIP number format failing. Calling log of my account indicated that failing calls was decoded and directed as calls to New Zealand instead of US calls. I suspect that N800 2008 SIP client doesn't send VOIP domain info and system can't properly decode destination.
Any thoughts?
Another problem with dialer - it is disabled after sending phone number until hung up. In siltation when caller needs to call phone extension no dialer available.
Can you guys confirm problems or show errors on my side?

Re: sip client os2008
 

in general im finding then nat traversal using default setting in sofia are working fine for me with two accounts (voipuser.org and sipgate.co.uk) but i've also found that sip domain diailing fails (sip-id@domain.com) and hope that this is resolved in the future as it's one of the ways in which i call friends on different sip providers even when they don't have reciprocal arrangements.

Re: sip client os2008
 

It seems that this bug affects other providers too (www.voipdiscount.com and www.voipbuster.com have been reported).

please vote in bugzilla if this concerns you.

Chris

Re: sip client os2008
 

SIP URIS work fine here, here's two that work for me:
1234@loligo.com
613@fwd.pulver.com

Do you have any luck with those?

I did have a problem initially, but that was due to my Asterisk server not being configured to dial anything other than regular phone numbers, or local extensions.

Re: sip client os2008
 

I had a similar problem with using sofia and gizmoproject SIP service.

To fix it I forced the Transport (Accounts / Edit / Advanced) to use just UDP instead of Auto.

Hope that helps.

p.s. Has anyone got a way to find / fix the RTP port ?