setting up obi110 to terminate my call and register it to obitalk without phone
joe123:
anyone?
joe123:
Quote from: azrobert on December 08, 2015, 08:20:25 am
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?
Quote
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?
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