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Posts: 61 | Thanked: 1 time | Joined on Apr 2007
#11
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
 
Mara's Avatar
Posts: 1,310 | Thanked: 820 times | Joined on Mar 2006 @ Irving, TX
#12
Originally Posted by tfinnan View Post
The internet is connected via a wifi router that is actually mounted below my desk here in the office.
This might be it... Is your WiFi router connected directly to your ISP or another HUB/router in your office? In order this to work the WiFi router has to have valid EXTERNAL IP address, accessible from outside internet.

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... 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!

What brand and model is your WiFi router?

EDIT: The routing really means "port forwarding" in this case... Sorry if this caused confusion...

Last edited by Mara; 2007-07-23 at 21:12.
 
Posts: 61 | Thanked: 1 time | Joined on Apr 2007
#13
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?
 
Posts: 168 | Thanked: 51 times | Joined on Jun 2007
#14
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.
 
Mara's Avatar
Posts: 1,310 | Thanked: 820 times | Joined on Mar 2006 @ Irving, TX
#15
Originally Posted by Mara View Post
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...
Tried this and still no go: The IP address on sipphone proxy server is not set up correctly.

Can anyone confirm if yours is working correctly when checked from the sipphone Advanced Features tab?
 
Posts: 168 | Thanked: 51 times | Joined on Jun 2007
#16
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.
 
Mara's Avatar
Posts: 1,310 | Thanked: 820 times | Joined on Mar 2006 @ Irving, TX
#17
Originally Posted by coffeedrinker View Post
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.
coffeedrinker: The transport protocol tab change to UDP did do the trick! Now I called my home SIP phone and I could hear the voice going from N800 to home SIP hardware phone. Still the voice did not get from home phone to N800 though.

Also the N800 SIP client is now logged in properly (correct IP address) to sipphone server.
 
Posts: 1 | Thanked: 0 times | Joined on Jul 2007
#18
Originally Posted by tfinnan View Post
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
Hi,


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
 
Posts: 373 | Thanked: 56 times | Joined on Dec 2005 @ Ottawa, ON
#19
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.
 

The Following User Says Thank You to mwiktowy For This Useful Post:
Posts: 193 | Thanked: 41 times | Joined on Jan 2007 @ Paia, Maui, Hawaii
#20
Originally Posted by mwiktowy View Post
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.
Running 'Pidgin' at the same time might help in the meantime.

Last edited by ichmoimeyo; 2007-07-30 at 10:07.
 
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