setting up obi110 to terminate my call and register it to obitalk without phone

[joe123]

hello guys,

i ve bought recently an obi110 for terminating my calls while i m outside.

so i have several questions:

1/is it possible to attache the 0bi110 to my obitalk account without using a phone?
if yes what is the process to do so?

2/what is the settings i need to use to terminate all my calls from the obion apps thru my landline?

thanks in advance

[drgeoff]

1. No, you need to use a phone to dial the **5 xxxx number to register your OBi with the portal.

But registering with the portal may not be required.  Except for configuring GV on your OBi you can do virtually everything else using the device's onboard web server GUI.

What exactly do you mean by "terminate"?

[joe123]

hello

ok so i ll need to borrow or buy a phone for this

terminate means i want all my calls going from my app obion to the obi110 connected to my landline.basically the landline will terminate all my calls.

like:

obion----obi110----landline pstn

so do you know how can i do this without adding the device in the obitalk?

[azrobert]

I suggest borrowing a phone. It might not work without registering your OBi110, but even if it did, calling from OBiON would be a pain. You wouldn't be able to specify your OBi110 as the gateway for OBiON, so you would have to include the OBi number for every call. Without getting into details, you would have to dial the number like "2*18005551212".

Be aware that OBiON for Android hasn't been updated since 2011 and some people with newer versions of Android have problems. You might have better luck with OBiON for iPhone.

[azrobert]

There is a more difficult setup using the CSipSimple softphone on an Android that would work without registering your OBi110.

[joe123]

There is a more difficult setup using the CSipSimple softphone on an Android that would work without registering your OBi110.

hi azrobert

can you explain me the process?

or do you have a link for that?

thanks

[azrobert]

Create a free Sip2Sip account here:
https://mdns.sipthor.net/register_sip_account.phtml

Install CSipSimple on your Android.

In CsipSimple:
Select Add Account
Select the Basic Wizard. It's near the end of the list.

Account Name: OBi110
User: Sip2Sip_ID
Server: sip2sip.info
Password: Sip2Sip_PW
Save

Select Menu on phone
Select Settings
Select Filters
Select your newly created Account
Select add filter
Select 1st entry (Can't call) and change it to Rewrite.
Select 2nd entry (Starts with) and change it to All.
Select 3rd entry (Replace...) and change it to Suffix With.
4th entry: @xx.xx.xx.xx:5060
Save

Sip2Sip_ID and Sip2Sip_PW are the name and password you selected when you created your account.
xx.xx.xx.xx is your public IP address assigned to your modem by your ISP.
5060 is the UserAgentPort of SP1.
You MUST dial from the Android Dialer and not the CSipSimple Dialer.
Filters won't work from the CSipSimple dialer.

Log directly into the OBi using the local interface to make config changes.
Key the IP address of the OBi into a Web Browser.
Hit Enter
The UserID and default Password are both "admin".

Click on the categories on the left column.
Uncheck box to right of value to be able to make changes.
Make all the changes for that page then click Submit on the bottom.
Do the same for next page.
When all changes are complete, click Reboot on top.

Configure OBi110
Service Providers -> ITSP Profile A -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile A -> SIP -> X_SpoofCallerID: Checked
Voice Services -> SP1 Service -> AuthUserName : OBi110
Voice Services -> SP1 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP1 Service -> X_InboundCallRoute:
{Sip2Sip_ID>(XX.):li}

Physical Interfaces -> line Port
InboundCallRoute: sp1(Sip2Sip_ID@sip2sip.info)

You need to port forward 5060 in your router to the OBi's IP address.
IF you have audio problems you will also have to port forward RTP ports 16600 thru 16798.
That's 199 ports.
See: Service Providers -> ITSP Profile A -> RTP -> LocalPortMin/Max
Use Port Range Forwarding to avoid having 199 entries in you router.

Any dialed number will be routed to PSTN
Inbound calls will ring your Android.

I already had this written, but had to make changes for your requirements. Hopefully I didn't make any mistakes.

[joe123]

thanks azrobert

i ve borrowerd a phone and trying to add the device to obitalk:

i m hearing the message and then hanging up,but nothing is happening after that.

any idea how to solve this?

thanks

[azrobert]

In OBiTalk do you see a count down in seconds and does it timeout?

The OBiTalk network sometimes has problems. Just keep trying or maybe wait until tomorrow.

[joe123]

hi

i ve been able to add it in obitalk

now what is the settings to make it possible to use from obion?

for routing all my calls thru my landline?

i ve succeeded in activating the attendant, but it s asking me to choose new call and all.

and i want my calls go thru pstn directly,what should i change in my settings?


thanks in advance

[joe123]

anyone?

[joe123]

Create a free Sip2Sip account here:
https://mdns.sipthor.net/register_sip_account.phtml

Install CSipSimple on your Android.

In CsipSimple:
Select Add Account
Select the Basic Wizard. It's near the end of the list.

Account Name: OBi110
User: Sip2Sip_ID
Server: sip2sip.info
Password: Sip2Sip_PW
Save

Select Menu on phone
Select Settings
Select Filters
Select your newly created Account
Select add filter
Select 1st entry (Can't call) and change it to Rewrite.
Select 2nd entry (Starts with) and change it to All.
Select 3rd entry (Replace...) and change it to Suffix With.
4th entry: @xx.xx.xx.xx:5060
Save

Sip2Sip_ID and Sip2Sip_PW are the name and password you selected when you created your account.
xx.xx.xx.xx is your public IP address assigned to your modem by your ISP.
5060 is the UserAgentPort of SP1.
You MUST dial from the Android Dialer and not the CSipSimple Dialer.
Filters won't work from the CSipSimple dialer.

Log directly into the OBi using the local interface to make config changes.
Key the IP address of the OBi into a Web Browser.
Hit Enter
The UserID and default Password are both "admin".

Click on the categories on the left column.
Uncheck box to right of value to be able to make changes.
Make all the changes for that page then click Submit on the bottom.
Do the same for next page.
When all changes are complete, click Reboot on top.

Configure OBi110
Service Providers -> ITSP Profile A -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile A -> SIP -> X_SpoofCallerID: Checked
Voice Services -> SP1 Service -> AuthUserName : OBi110
Voice Services -> SP1 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP1 Service -> X_InboundCallRoute:
{Sip2Sip_ID>(XX.):li}

Physical Interfaces -> line Port
InboundCallRoute: sp1(Sip2Sip_ID@sip2sip.info)

You need to port forward 5060 in your router to the OBi's IP address.
IF you have audio problems you will also have to port forward RTP ports 16600 thru 16798.
That's 199 ports.
See: Service Providers -> ITSP Profile A -> RTP -> LocalPortMin/Max
Use Port Range Forwarding to avoid having 199 entries in you router.

Any dialed number will be routed to PSTN
Inbound calls will ring your Android.

I already had this written, but had to make changes for your requirements. Hopefully I didn't make any mistakes.

hi azrobert
i ve tried this but my android doesn t allow me to choose the obi110 filter as it s in grey while showing the choice

any idea how to make it work?

edit:was able to dial,but i m getting message saying calls to pstn are forbidden,it seems sip2sip.info is blocking the calls going thru? or am i doing something wrong?

thanks

[joe123]

it seems sip2sip is blocking pstn calls


i ve tried the same setting with iptel.org

but when i m dialing it s only ringing the phone attached to the obi110 ,not going thru pstn

what should i change?

edit 2:calls are now going thru but there is no voice now

[azrobert]

Are you port forwarding the RTP ports as described in my post?

IF you have audio problems you will also have to port forward RTP ports 16600 thru 16798.
That's 199 ports.
See: Service Providers -> ITSP Profile A -> RTP -> LocalPortMin/Max
Use Port Range Forwarding to avoid having 199 entries in you router.

Is this the only problem you have with CSipSimple?

To fix OBiON:
Voice Services -> OBiTalk Service -> InboundCallRoute:
{290123456>(XX.):li},{ph}

Change 290123456 to the OBi number of your OBiON app.

[joe123]

it s working fine with obion

and yes i ve forwarded all the ports in my modem to point them to the obi110 ip.

any ideas?

[azrobert]

it s working fine with obion

Great!

any ideas?

If I understand correctly, the call completes but you have no audio.

Make a call.
Sign into OBi Expert
Click on Status on the left column
While the call is active, click on Call Status
It should show both legs (SP1 and Line) of the call
Do you have rx and tx packet counts for SP1?
What are the codecs for tx and rx for SP1?
Is the Local RTP Address port number one of the ports you port forwarded?

Are you using a local WiFi connection or your cell network.
If it's WiFi, try the cell network.

[joe123]

hi azobert

sorry i didt take time to use your settings

so i wasnt able to use sip2sip as they are blocking pstn.

so i switched the settings to iptel and same issues it s ringing but there is no voice going thru

the codec in tx is g711 u and the port forwarded is going to obion ip ,but there is  rx codec and there is no rtp nor in rx nor in tx...

so what can be the issue?

thanks and sorry for the delay