Bridging to SIP doesn´t work

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Montbra:
Hi to everyone.

I´m trying to to bridge all my Line Inbound Calls to my Sip2Sip account and use my smartphone like a remote home extension over 4G using my Obi110. I've been reading lot of posts and special Ianob and Azrobert stuffs, making a dummy sp2 service and setup Line Inbound Calls like this to test:

InboundCallRoute = {sp2(xxxxxx@sip2sip.info;ui=$1),pp(obxxxxxxxxx),ph}
 
This way, to test, I can ring my phone (obvius) and the Obion app, but not my Zoiper configured to sip2sip.

I did a test with my other smartphone and setup a Getonsip account and be able to make sip calls on both each other than I presume this part is ok.

I´m run out of ideas to resolve this.

Anyone can help?

Thanks in advance and best regards from Rio de Janeiro, Brazil.
PS: Sorry to any english errors.

azrobert:
Look at the OBi110's call history for any errors or a reason for failure.

To access Call History:
Log directly into the OBi110 using the local interface.
Key the IP address of the OBi110 into a Web Browser.
Hit Enter
The UserID and default Password are both "admin".
Click Status on the left column.
Then click Call History.

Are you sure you defined the dummy SP2 trunk correctly?
It should look like this:
Service Providers -> ITSP Profile B -> SIP -> ProxyServer: 127.0.0.1
Voice Services -> SP2 Service -> AuthUserName: anything
Voice Services -> SP2 Service -> X_RegisterEnable: unchecked
Voice Services -> SP2 Service -> X_ServProvProfile: B

If SP1 is currently defined as an SIP service, you can use it:
{sp1(xxxxxx@sip2sip.info;ui=$1),pp(obxxxxxxxxx),ph}

Quote

I did a test with my other smartphone and setup a Getonsip account and be able to make sip calls on both each other than I presume this part is ok.
I'm not sure what this means.
Are you able to call the smartphone registered to Sip2Sip from a smartphone registered to GetOnSip?

Montbra:
Hi Azrobert, thanks in advance.

My dummy sp2 service is set like your recommendation (GV in SP1) and, yes, I´m able to call my phone registered in GetOnSip to my other phone with Sip2Sip account and vice versa, than I presume that part is correctly set. This is burning my mind.

Later at night I will verify this call log and make some tests, because my set up is in my home and let you know. Anyway, can you explain de difference between X_ServProvProfile: A and B?

Again thanks for help.

azrobert:
Quote from: Montbra on July 03, 2015, 09:00:55 am

Anyway, can you explain de difference between X_ServProvProfile: A and B?

You define a trunk with a Service Provider entry and a Voice Services entry.
The Service Provider section basically defines the Proxy.
The Voice Services section defines the UserID info.
The X_ServProvProfile points the Voice Services section to a Service Provider.

Assuming you have 2 accounts at Sip2Sip.
You define Service Provider A Proxy as sip2sip.info and Voice Services SP1 as user1.
You define Voice Services SP2 as user2.
Since both accounts use the same Proxy, you can leave X_ServProvProfile at "A".
Now you don't have to define a Service Provider B.

I don't think many people do this, so I don't know why the default is "A" on all the X_ServProvProfiles.

If you didn't change the SP2 X_ServProvProfile to "B", SP2 will be defined as GV and SIP calls won't work.

Montbra:
Azrobert, now I understand A and B. Thanks.

Anex is a print from my Call History. SP2 without parameters. SP2  is configured with my Sip2Sip account. Is this correct?

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