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Settings for gizmo SIP?
Does anybody know what are the settings for SIP(using the n800) if using the sip provided by gizmo project. WOOHOO for their incoming number
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Re: Settings for gizmo SIP?
This information would be helpful, as I am attempting to setup the new SIP client as well, and would like to figure out how it works a bit better.
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Re: Settings for gizmo SIP?
It is written right on the rtcomm site.
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Re: Settings for gizmo SIP?
Maybe I'm missing something, but are sipphone.com and Gizmo the same thing? If this is the case, then I understand what you mean about the settings being on rtcomm's site. I tried the settings that were listed under that section, to no avail. If they aren't the same, I didn't see Gizmo mentioned anywhere at all on the rtcomm site.
I have read just about everything that I can find on both ITT and in rtcomm's garage site, and can't quite get this going. Any assistance would be appreciated. I am also attempting to get Truphone to work with the now integrated SIP service. I saw a post on another site alleging that this was possible, but haven't been able to find it again. Again, any direction would be appreciated. |
Re: Settings for gizmo SIP?
This should help:
http://support.gizmoproject.com/inde...ndetails&_i=88 |
Re: Settings for gizmo SIP?
If you bought a number at gizmo, you are using sipphone.com as far as I know. You can check this by trying to login with your gizmo login credentials at sipphone.com. They work perfectly fine both sending and receiving calls. If you're still lost later tonight, I'll post exactly what my settings are.
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Re: Settings for gizmo SIP?
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I did try to change the ports, STUN servers, etc... on the N800 SIP client with no avail. To me it looks like the STUN server settings are not yet supported on the N800 SIP client. |
Re: Settings for gizmo SIP?
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EDIT: I just got an idea... I have another Hardware SIP phone adapter logged in SIPPHONE proxy server and it uses the same STUN server. It could be that my router routes the STUN server return packets to my HW adapter, thus the N800 will not get its STUN reply... I'll try this later again by disconnecting the HW adapter from my network to see if it works then... |
Re: Settings for gizmo SIP?
Okay. First and foremost, thanks to all of you guys (zerojay, mara, and iball). I didn't really expect to get much help with this query. Here's where I stand right now:
No additional apps running in the background. When I attempt to make a call, I get a 'unable to establish connection' error message. The internet is connected via a wifi router that is actually mounted below my desk here in the office. The amplified antenna for the router is mounted on a shelf near my desk, so there is no issue with signal strength. My settings as they currently exist are as follows: User name: <my Gizmo number> @ 'proxy01.sipphone.com Password: <my Gizmo password> Account name: Gizmo On Advanced Settings Tab/Connection Transport is set to: Auto Outgoing Proxy: proxy01.sipphone.com (I also experimented with leaving this blank) Port: 5060 Discover Public Address: <checked> Keep Alive Mechanism: Auto Keep Alive Frequency: 0 (that was the default, so I didn't change it) On Authentication Tab/Authentication Username: <blank> (this is blank now, but I also experimented with entering my username and password in the appropriate spaces on this page) Password: <blank> On STUN Tab/Auto Detect Stun: <unchecked> STUN Server: stun01.sipphone.com (I have also tried with this blank and 'Auto-Detect STUN checked) STUN Port: 3478 I apologize for the length of this post, but this is exactly how I have the account set up right now, and I have not been able to connect. If you get the opportunity, and see any flaws in the way that this is set up, please don't hesitate to let me know. I'm certain that I am not the only person who has had problems with setting this up, and hope that this might help someone (who might not have asked) to get this right as well. Thanks again, and let me know if you see anything. -Tim |
Re: Settings for gizmo SIP?
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Then it is another story if your router can do the routing properly... at worst you may need to set up your N800 for static IP address and set up static routing table into your router... :eek: If there are more routers in your network before reaching your WiFi router then each router needs to have their routing tables set up properly! :eek: :eek: :eek: What brand and model is your WiFi router? EDIT: The routing really means "port forwarding" in this case... Sorry if this caused confusion... |
Re: Settings for gizmo SIP?
Mara.
The router is just a cheap Linksys. I am unsure of the number off the top of my head. This shouldn't be the issue, however, as I am able to connect via the regular Gizmo client on both my N800 and my 770, Skype on the N800, and via Truphone and Gizmo on my N80. We are talking about my home office, so the router is directly connected to my ISP. The other wireless router (upstairs and on the other side of the house) is also connected directly to my ISP. Both routers work with the aforementioned SIP (or semi-sip) applications. Would it stand to reason that the other applications wouldn't work if the port forwarding wasn't set up properly? |
Re: Settings for gizmo SIP?
Had the same problem with sound not working but the support page has the answer:
Tips and tricks Using sipphone.com service 1. Sipphone SIP service does not support chat. 2. When configuring an account: - enter proxy01.sipphone.com as a domain - enter stun01.sipphone.com as a stun server(advanced settings) - select UDP transport (advanced settings) That got it working for me. |
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Can anyone confirm if yours is working correctly when checked from the sipphone Advanced Features tab? |
Re: Settings for gizmo SIP?
My full settings (that work):
Window: Account Setup (don't put dashes in your phone number) User Name: 174xxxxxxxx @ proxy01.sipphone.com Password: gizmoprojectpassword Advanced settings: Connections Tab: Transport: UDP Port: 5060 (default) all the others are defaults Authentication Tab: empty STUN Tab: STUN server: stun01.sipphone.com STUN port: 3478 I use OpenDNS for name service. I am behind a chain of THREE DLink routers. |
Re: Settings for gizmo SIP?
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Also the N800 SIP client is now logged in properly (correct IP address) to sipphone server. |
Re: Settings for gizmo SIP?
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We (RTComm team) are looking into these problems atm... Could you please file a bug on maemo's bugzilla describing your case (https://bugs.maemo.org/enter_bug.cgi?product=rtcomm)? It'll be much easier for all of us to track and solve the problem then. -- Andrei Laperie, RTComm team |
Re: Settings for gizmo SIP?
One thing that Gizmo does is use jabber for the presence and contact info.
So to chat with other Gizmo users and show up as online for them, a separate jabber account needs to be set up (gizmo-user-name@chat.gizmoproject.com). Unfortunately, there doesn't seem to be a way yet to bind the jabber and sip accounts together like the gizmo client does. |
Re: Settings for gizmo SIP?
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Running 'Pidgin' at the same time might help in the meantime. |
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So while Pidgin is a nice piece of s/w ... it doesn't help here. |
Re: Settings for gizmo SIP?
Hey guys, great thread! I followed the info in Coffeedrinkers post, but before that I installed the rtcomm beta. It seemed to work out, and the green light at the top of the n800 screen in OS2008 would flash green/red and then stay solid green. Great, so I open chat and dial a call. I tried a few different numbers and a few different ways of entering the number, ie "18005551212" and "8005551212." The sip icon is there, and it starts ringing, but it just rings and rings, but never completes the call. Am I doing something wrong in dialing? For the heck of it, I downloaded the gizmo client and was able to make calls on the n800 to land lines with out a hitch, the other party could hear me and all. But I'd rather dial with my sipphone account through the chat app. Thanks!
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