Bridging to SIP doesnīt work
azrobert:
The Call History doesn't look correct. I duplicated your setup on a test OBi110 and my call history shows the full URI address after SP2.
I couldn't get your previous configuration to work. The below configuration works for me. Do you need Sip2Sip defined on SP2?
Service Providers -> ITSP Profile B -> SIP -> ProxyServer: 127.0.0.1
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID: Checked
Voice Services -> SP2 Service -> AuthUserName: anything
Voice Services -> SP2 Service -> X_RegisterEnable: unchecked
Voice Services -> SP2 Service -> X_ServProvProfile: B
SP1 Inbound route:
{sp2(xxxxxx@sip2sip.info),pp(obxxxxxxxxx),ph}
If it still doesn't work try a speed dial defined like:
sp2(xxxxxx@sip2sip.info)
What does the call history show?
Montbra:
I didn't explain this correctly. I didn't setup sp2 with my sip2sip account. I did a dummy server like you explained. I tried to pass the uri to SP2 but didnīt work. I don't understand this part:
SP1 Inbound route:
{sp2(xxxxxx@sip2sip.info),pp(obxxxxxxxxx),ph}
because my intention is fork a line inbound call. What is your intention here?
I put sip uri in speed dial position 1 but I don't know how to use it. Is it like sp2(1)?
Montbra:
My fault, I understand now. I dial 1# in my phone and ring in my sip2sip account in smartphone. Working this way.
Sorry to be so noob
azrobert:
Are inbound calls on SP1 successfully bridged to your smartphone?
Bridged calls in the Call History don't look correct.
It should look like this:
Forking to: SP2(xxxxxx@sip2sip.info), OBiTALK1(ob290xxxxxx), Phone1)
I made the following changes to get mine to work:
I removed the ";ui=$1" from the inbound route.
This is used to pass CallerID and I think it was causing a problem.
SP1 Inbound route:
{sp2(xxxxxx@sip2sip.info),pp(obxxxxxxxxx),ph}
I then enabled X_SpoofCallerID.
This is another method to pass CallerID.
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID: Checked
If you can't get the bridged calls to work, I suggest using Callcentric instead of Sip2Sip. Callcentric has a free basic account called an IP Freedom account. You can sign up here:
http://www.callcentric.com/rate/plans/ip_freedom/
Let me know if you want to use Callcentric and I'll walk you thru the setup.
Montbra:
Azrobert, forking to SP1 doesn't work either. I tried put @ in front the rule without success. I tried only SP1 in rule and receive this: End Call (482 Loop Detected), see picture in annex. I'd like to try the CallCentric solution, but this fault is disturbing my mind. Is a question of honor resolve it.
Best Regards
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