Bridging to SIP doesn´t work

[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}

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]

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?

[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 didn´t 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

[azrobert]

forking to SP1 doesn't work either

I suggested using SP1 before I knew it was defined as GV. The trunk you bridge the call to must be defined as SIP, so SP1 won't work.

Is a question of honor resolve it.

I agree. Also, you might have the same problem with Callcentric.

The Call History should look like this:
Forking to: SP2(xxxxxx@sip2sip.info), OBiTALK1(ob290xxxxxx), Phone1)

Not:
Forking to: SP2(), OBiTALK1(ob290xxxxxx), Phone1)

Something is drastically wrong.
Please check your firmware build level.
In OBi Expert go to Status -> System Status
Under Product Information what is the SoftwareVersion?
I just need the Build number.

[Montbra]

Something is drastically wrong.
Please check your firmware build level.
In OBi Expert go to Status -> System Status
Under Product Information what is the SoftwareVersion?
I just need the Build number.

From local Web Interface:
Hardware version 3.4
Software Version   1.3.0 (Build: 2872)
***6 answer that update is not available

[azrobert]

For a test try:
sp2(xxxxxx@sip2sip.info)

This will show any error. When forking errors might not show.

Edit:
We can test Callcentric in parallel with sip2sip.

http://www.callcentric.com/products/
Click Order Now to the right of IP Freedom
Enter New Customer Info

Configure Softphone with another account
Proxy: callcentric.com
User: 1777xxxxxxx
Password: CC_PW

OBi110 inbound route:
sp2(1777xxxxxxx@in.callcentric.com)

[Montbra]

For a test try:
sp2(xxxxxx@sip2sip.info)

This will show any error. When forking errors might not show.

This way speed dialing work and not forking, using only sp2 with sip2sip account doesn't work. Call 4 and call 3 respectively. Note that call 3 doesn´t has peer number.

Edit:
We can test Callcentric in parallel with sip2sip.

http://www.callcentric.com/products/
Click Order Now to the right of IP Freedom
Enter New Customer Info

Configure Softphone with another account
Proxy: callcentric.com
User: 1777xxxxxxx
Password: CC_PW

OBi110 inbound route:
sp2(1777xxxxxxx@in.callcentric.com)

Success! This way work in speed dial and forwarding to sp2 CallCentric account. Call 2 and call 1. In this case peer number appear in both. Caller ID doesn´t spoofed but is checked.

Question is, why?

[azrobert]

1st, I misunderstood what you wanted to do. I thought you wanted to bridge inbound SP1 calls, not Line calls. I don't know why because you clearly stated Line calls in your original post. Anyway, no harm done because you knew where to place the code even when I said SP1 inbound route.

It looks like everything is working correctly with Callcentric, so let's forget about Sip2Sip. Do you need your cell to be able to call your OBi110? In Callcentric you would setup another extension, like 101. Change SP2 to register to Callcentric using AuthUserName=1777xxxxxxx101. On the cell call 101.

The CallerID problem is a Line configuration problem. Look at the call history. Peer Name should contain CNAM and Peer Number should contain CallerID, so the OBi110 is not recognizing these on inbound Line calls.

Are you getting CallerID on the phone attached to the OBi110? If you are getting CallerID then you probably changed RingDelay on the Line Port configuration page. RingDelay is required for bridged calls to pass CallerID. I just remembered you are not in N. America. The above is true in N. America because the CallerID is sent between the 1st and 2nd ring. I don't know if this is true in Brazil.

The other parameter to look at is CallerIDDetectMethod.  

If the above doesn't work, I don't know what else to suggest. Maybe someone else can help.

Edit:
The default RingDelay will cause a 4 second delay before the Phone port will ring. If in Brazil the CallerID is sent before the 1st ring, you can set the RingDelay to 0

Edit2:
Line Inbound route:
{sp2(1777xxxxxxx@in.callcentric.com),pp(obxxxxxxxxx),ph}