when one has more than one sip registration from the e61 (I have 3 simultaneous
voip providers so I can receive calls on all) how does one select a particular
one to make an outgoing call? I have to de-register all but the one I want to
use for the outgoing internet call.. could you perhaps start a topic on this?

By Tezza

In tools>settings>connection>internet tel settings>vbuzzer>
name: vbuzzer
SIP profiles: vbuzzer

Sip Setings>vbuzzer (not registered)
Settings:
prifile name – Vbuzzer
service profile – IETF
Default access point – my own access point name
public user name – sip:my user name@vbuzzer.com
use compression – yes
registration – when needed
use security – no

Proxy Server
proxy server address – sip:vbuzzer.com
realm – vbuzzer.com
user name – my user name – “xxxxxxxxxx”
password – “xxxxxx”
allow loose routing – yes
transport type – UDP
port – 80

Registrar Server
proxy server address – sip:vbuzzer.com
realm – vbuzzer.com
user name – my user name – “xxxxxxxxxx”
password – “xxxxxx”
transport type – UDP
port – 80

WIRELESS LAN
Show availability – yes
Scan for networks – every 5 minutes

MENU>CONNECTIVITY>INTERNET TEL- “DISCOVERING SERVICES-on top of the screen, it says:vbuzzer not connected, and if I click my access point below, – it connects to vbuzzer via my access point and continue to do so – nothing happens, then it says, “unable to connect”

SETTINGS>CALL>
Internet call waiting – activated
Internet call alert – on
Dafault call type – internet

Thank you all. If we can make this work, then we can experience full enjoyment of the E61 and VOIP-Vbuzzer and share with others in similar situations.

 Anyone have any working Wengo settings they could share?

1) How do i make use of GPRS to make cheap calls? Any software i have to download to my phone?

2) If I connect my E61 to my internet enabled PC @ bandwidth of approx 42kbps through the data cable, how do I make use of my E61 to make calls using the internet and not GPRS?

1. I choose “pbxes.com” it’s a German VOIP provider of free VOIP PBX what means that you can setup your personal Online Telephone Station.

2. We need a “trunk” a gateway for making calls to ordinary phones worldwide. It will be “Gizmo Project”.

Now you need to register on both sites and create accounts on them just follow there guides it’s easy so I’m goning to pass this. Now after you created accounts lets setup everything up.

1. Login to your “pbxes.com” account and goto “Extensions” -> “Add Extension” -> choose “SIP”.

1a. In “Extension Number” enter “100″.
1b. In “Display Name” enter “My e61 phone”.
1c. In “password” enter any password, let’s say “testing”.
1d. Press “Submit”.

2. Now goto “Trunks” -> choose “Add SIP Trunk”.

2a. In “Trunk Name” enter “sipphone”.
2b. In “Account – username” enter your “Gizmo Project” account, example “1747*******”.
2c. In “Account – password” enter your “Gizmo Project” password.
2d. In “Account – SIP Server” enter “proxy01.sipphone.com”.
2e. Press “Submit”. That’s it other fields just leave as they are do not need to fill them in.

3. Now goto “incoming Calls”.
3a. In “regular hours” and in “after hours” select “Extension” and choose your extension, should be “100″ as we had it setup already.
3b. Hit “submit”.

4. Now goto “Inbound Routing” -> choose “Add Incoming Route”.
4a Select “Extension” and choose your extension, should be “100″ as we had it setup already.
4b. Press “Submit”.

5. Now goto “Outbound Routing” -> choose “Add Route”.
5a. In “Route Name” enter “sipgate”.
5b. In “Dial Patterns” enter “XX.” don’t forget a dot after XX.
5c. In Trunk Sequence select “SIP/sipphone”
5d. Hit “Submit”.

To verify ethat verything works goto “Status”.
1. Extension “100″ and Trunk “proxy01.sipphone.com” should be green. If they red you’ve done mistake somewhere.

You are halfway done, now let’s setup e61 now with new VOIP SIP settings.

1. On Nokia e61 goto “Tools” -> “Settings” -> “Connection” -> “SIP settings” -> hit options and choose “Add new” -> “Use default profile”.
1a. In “Profile name” enter “pbxes”.
1b. In “Service profile” choose “IETF”.
1c. In “Default access point” choose your connection. (it should be “WiFI” with GPRS I cannot guarantee quality).
1d. In “Public user name” enter “sip:yourPBXESusername-100@pbxes.org” (note. type your username you registered with PBXES).
1e. In “Use compression” select “NO”.
1f. In “Registration” select “Always ON” you may choose other later.
1g. In “Use security” select “No”.

2. Continue. Now in “Proxy server”.
2a. In “Proxy server address” enter “sip:217.195.32.11″.
2b. In “Realm” enter “pbxes.org”.
2c. In “User name” enter “sip:yourPBXESusername-100″ (note. please don’t forget to put minus “-” between extension and username).
2d. In “Password” enter the password you created for your extension in our case it is “testing”.
2e. In “Allow loosing routing” select “Yes”.
2f. In Transport type choose “UDP”.
2g. In “Port” should be “5060″.

3. Now hit “Back” and goto fill “Registrar server”.

3a. In “Registrar server address” enter “sip:217.195.32.11″.
3b. In “Realm” enter “pbxes.org”.
3c. In “User name” enter “sip:yourPBXESusername-100″ (note. please don’t forget to put minus “-” between extension and username).
3d. In “Password” enter the password you created for your extension in our case it is “testing”.
3e. In Transport type choose “UDP”.
3f. In “Port” should be “5060″.

Now you almost done now quit and goto:
1. “Tools” -> “Settings” -> “Connection” -> “Internet tel. settings” -> hit options and choose “New profile”.
1a. In “Name” enter “pbxes”.
1b. In “SIP profiles” choose “pbxes”.

On the very right of your status bar on the phone you should see “phone icon with earth” to apear.

Now you done to test your connection dial “*43″ hit “Options” chose “Call” -> “Internet call”.

You will here “Echo test”. As I know when you signup with “Gizmo” they give you Free 10 seconds so you my try to call someone on a landline use this combination when you dial “+(countrycode)(areacode)(phonenumber)” without “()”.

Well “pbxes.com” offer much more interesting features like redirecting your internet calls to your mobile phone when you out of WiFi coverage and like smart Rount selecting depending on where you call. The WEB service is based on well know Asterisk open source PBX.

If you got intrested on this topic let me know and I’ll continue to describe wonderful features of SIP VoiP.