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Posts: 2 | Thanked: 1 time | Joined on Jan 2008
#1
Hello all:
I used to be able to set up SIP account(from voipcheap.co.uk) on my n800 with os2007. But after upgrade to OS2008, I can still get connected to the SIP( and able to receive calls) but not able to make a call. An error message says: "Not able to establish connection".
Does anyone has the similar problem? and happens to know how to solve it?

cheers,
 

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Posts: 124 | Thanked: 34 times | Joined on Mar 2007
#2
On the topic of SIP problems on N800 with OS2008, I signed up with Gizmo project, and I've successfully made outbound gizmo calls, but I haven't ever received one - just a lot of missed-call notifications.

Just to see if it changed anything for me, I tried using RTCOMM's internet phone support to try to make/receive SIP calls, and no dice. In fact, it seems to cause the gizmo client on my brother's 770 and his desktop to crash when I call using RTCOMM.

I'm most interested in getting RTCOMM working, because then I don't have to have Gizmo running all the time in order to receive calls.

Here's my account setup information:
[Wizard]
User name: (my gizmo login username) @ proxy01.sipphone.com
Password: (duh)
Use for PSTN calls: [ ]

Account Name: Sipphone

[Advanced - Connection]
Transport: Auto
Outgoing proxy: ___________ (blank)
Discover public address: [+]
Keep-alive mechanism: Auto
Keep-alive frequency: 0

[Advanced - Authentication]
Authentication user name: _____________ (blank)
Password: _____________ (also blank)
(how is this username/password different from the ones I gave in the wizard?)

[Advanced - STUN]
Auto-detect STUN: [+]
STUN server: __________ (grayed out)
STUN port: [3478] (grayed out)
After adding SIP account information to one of my pre-existing contacts, it now shows two sip accounts that I can call:
gizmo_username@proxy01.sipphone.com [SIP] _______ (blank)
gizmo_username@proxy01.sipphone.com [SIP] Sipphone

Why are there two, and are the both the same? Does it matter which one I use? Any suggestions as to why this isn't working?
 
Posts: 393 | Thanked: 112 times | Joined on Jul 2007
#3
Maemo bug tracker lists a SIP "bug" based on a low bandwidth codec.

Once disabled all was well on my end - search bug tracker - it's on there.
 
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