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Author Topic: Replace OBiON app - ObiON replacement - OBiON alternative  (Read 117226 times)
azrobert
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« Reply #20 on: March 03, 2014, 02:54:57 pm »

There are two problems with your port forwarding.
You have the protocol for 5061 as TCP.
It should be UDP.
The other 2 entries are correct with UDP.

In my router I have 2 different types of port forwarding.
The first you specify a single port to be forwarded to an IP.
The second type you specify a range of ports to be forwarded to an IP.
I don't see Port Range Forwarding as an option on your router.
Maybe you can specify a range of ports in the same area.

Here is the second problem.
It looks like you are forwarding 16800 and 16998.
I wanted you to forward 16800 thru 16998.
That's 199 ports.
DO NOT spend any time trying to resolve this.
Calls might work correctly without forwarding those ports.
Calls will complete even if there is a problem, but you will have one way audio or no audio.
Once you get this working and you have audio problem we can address it at that time.


Here is your new SP2 X_InboundCallRoute:
{pvpham>(Msp1):sp1},{101>17771234567:aa},{ph}

pvpham is your userid on Sip2SIP.
You still will be able to use Callcentric and the AA to make calls.

You can optionally make this change.
Service Providers -> ITSP Profile A -> DigitMap:
(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|<1480>xxxxxxx)

Change 480 to your local area code for 7 digit dialing.

One more thing to check in CSipSimple.
When making a call you must use the phone's native dialer.
For some reason filters don't work if you use the CSipSimple dialer.
You must integrate CSipSimple with the phone's dialer.
I don't remember the defaults in CSipSimple.

Check CSipSimple:
Select Settings
Select User Interface
Dialer Integration must be checked
I also have Calls Logs Integration checked

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pvpham
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« Reply #21 on: March 03, 2014, 06:40:48 pm »

azrobert,

I did the following:

change the TCP to UDP in one of my router port forwarding rule

change SP2 X_InboundCallRoute to:
{username>(Msp1):sp1},{101>7771234567:aa},{ph}

Change ITSP Profile A -> DigitMap to: (1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|<1941>xxxxxxx)

CsipSimple filter :
ALL
Sufix with
@myIP:port

Dialer Integration checked
Calls Logs Integration checked

Call direct not working. error 404 wrong number dial.
AA dialing still work.

Quote
You register CSipSimple to service Sip2SIP.
I did this too just like I registered CsipSimple with Callcentric


In the screen shot of jjjooonnn he has a change in SP1 X_inboundCallRouts:
(sp2(username@sip2sip.info),ph)

Do I need to change mine ?

Thanks



« Last Edit: March 04, 2014, 08:15:46 am by pvpham » Logged
azrobert
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« Reply #22 on: March 03, 2014, 09:02:13 pm »

Did you use the phone's native dialer and not CSipSimple's?

You can login to your Sip2Sip account and look at the call history to see if the call got there.

The SP1 inbound route change is to route inbound calls on SP1 to your cell phone. It is not needed for this to work.
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pvpham
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« Reply #23 on: March 04, 2014, 05:46:33 am »

Quote
Did you use the phone's native dialer and not CSipSimple's?
I'm sorry that I do not understand this.

I dialed by opening CSipSimple's Icon.
I checked the history at Sip2Sip: It's a blank page.

There is another phone Icon at the bottom of my phone( I guess this is what you means native dialer ?) I clicked on it just now and try dialing my wife cellphone, no tone no audio. I click on its history. It showed all the call I made using Csipsimple.

Thanks
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azrobert
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« Reply #24 on: March 04, 2014, 07:05:33 am »

Yes, use the phone icon on the home page and not CSipSimple's dialer.
After you dial the number an option window will pop up with two choices.
Sip2Sip or Mobile.
Under Sip2Sip will see the URI you are calling:
18005551212@xx.xx.xx.xx
Choose Sip2Sip or whatever you called the account you created.

The Sip2Sip history should have any entry even if you used the wrong dialer, so something is wrong.

Make sure CSipSimple is registered to Sip2Sip.
In CSipSimple select accounts.
This will show your Sip2Sip account.
Next to the account will be an icon.
Under the icon should be a green line.
Under the account name is should say "Registered".
If it's not registered tap the icon.

I unregistered CSipSimple on my phone and tried a call.
I did not get the pop up window and the call failed, so I think this is your problem.

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azrobert
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« Reply #25 on: March 04, 2014, 07:45:35 am »

PVPHAM,

I sent you a personal message.
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azrobert
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« Reply #26 on: March 04, 2014, 09:31:37 am »

If you can't find Port Range Forwarding there is a way to solve the problem. You don't need 199 RTP ports. Do this:

Service Provider -> ITSP Profile B -> RPT
LocalPortMax: 16405

Then port forward 16400, 16401, 16402, 16403, 16404 and 16405.

« Last Edit: March 04, 2014, 09:38:24 am by azrobert » Logged
pvpham
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« Reply #27 on: March 04, 2014, 04:13:45 pm »

azrobert,

I find the way to forward a range of port with my modem. after I did that I checked the Sip2sip history page. It showed record of my call but I could not use the information to diagnose my problem. Dialing from CsipSimple dialer has audio but still got error message 404 wrong phone number. Native dialer has no audio and return the same error like CsipSimple dialer.

Thanks
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azrobert
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« Reply #28 on: March 04, 2014, 07:17:27 pm »

The call history entry in Sip2Sip should look like this:
16235941000@xx.xx.xx.xx (audio)

Please post what you are getting.

When using the native dialer does it act like I described at the beginning of my reply#24?

Edit

If call history entry in Sip2Sip looks like this:
16235941000@sip2sip.info (audio)

Then the filter in CSipSimple is not working.
« Last Edit: March 04, 2014, 08:30:27 pm by azrobert » Logged
pvpham
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« Reply #29 on: March 05, 2014, 08:09:37 am »

Sip2Sip call history is as followowing:

16235941000@sip2sip.info (audio).

The filter I set for Csipsimple:

All
Suffix with @myIP address:SP2 port

Quote
Under Sip2Sip will see the URI you are calling:
18005551212@xx.xx.xx.xx
Choose Sip2Sip or whatever you called the account you created.


The format under Sis2Sip is: Call: xx.xx.xx.xx

Thanks

Patrick

Thanks

Patrick
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azrobert
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« Reply #30 on: March 05, 2014, 10:31:31 am »

When the History shows:
16235941000@sip2sip.info (audio).
The filter is not working or you used the CSipSimple dialer.
The above is trying to call user 16235941000 at Sip2Sip.

This looks good to me.
Quote
All
Suffix with @myIP address:SP2 port

This doesn't look right.
Quote
The format under Sis2Sip is: Call: xx.xx.xx.xx

I tried a few calls and my Sip2Sip call history seems broken.
It only shows calls to another Sip2Sip user.
It doesn't show calls to my OBi.
It was working on Monday.

Did you try dialing from the CSipSimple dialer?
Maybe calls from the native dialer are not showing in the call history and the entry is from the CSipSimple dialer.

Did you change the OBi port number?
All I can suggest is re-check everything you did.


ianobi or anybody,

Do you have any ideas?
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ianobi
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« Reply #31 on: March 05, 2014, 11:36:36 am »

I just read through this thread and I cannot see any obvious problem. My CSipSimple via sip2sip seems to test ok just now. Oddly, I'm also having trouble with the sip2sip website just now.

It might be worth putting the InboundCallRoute back to simply "ph". Then anything dialled from the cell phone with the suffix @myIP address:SP2 port should ring the OBi phone. This may narrow down where the problem is. If it works, then the problem must be the UserID or the number dialled.

I'll have a deeper look tomorrow (my time zone) and see if I can suggest anything further.


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pvpham
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« Reply #32 on: March 05, 2014, 06:28:38 pm »

azrobert and ianobi,

Thanks for your help and your patience

I came back and check everything and all is "exactly" as I am instructed to do. Well I read and re-read the comment by az below:

Quote
Under Sip2Sip will see the URI you are calling:
18005551212@xx.xx.xx.xx

Then I realize that I set the filter rule for Callcentric but I did not set the filter rule for Sip2Sip because under the "red bull eye " of Callcentric I see 18005551212@xx.xx.xx.xx. but under Sip2Sip I only see xxx.xxx.xxxx.

After I set the rule for Sip2Sip. It works like a champ.

I am so HAPPY.

Thanks you Az for helping me understanding what I'm doing and understand what should not post on line. It's been fun. It encourages me to explore and learn something new.
« Last Edit: March 05, 2014, 06:30:20 pm by pvpham » Logged
azrobert
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« Reply #33 on: March 05, 2014, 09:02:57 pm »

You are welcome.

Here is some more info.

To call the OBi's Phone Port from From CSipSimple dial a non-phone number like 100.

To call CSipSimple from the OBi setup a Speed Dial like:
sp2(pvpham@sip2sip.info)

To route inbound calls on SP1 to CSipSimple.

Change SP1 X_InboundCallRoute To:
{ph,sp2(pvpham@sip2sip.info)}

« Last Edit: March 05, 2014, 09:21:08 pm by azrobert » Logged
pvpham
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« Reply #34 on: March 06, 2014, 06:03:33 am »

azrobert,

Thanks for the info.

Quote
To call CSipSimple from the OBi

You means make a call using CSipSimple dialer?

Quote
setup a Speed Dial like:
sp2(pvpham@sip2sip.info)

Is this where I set the speed dial for the OBi: Users Settings>Speed dials>

Other question:

1) how do I send a text message from my smartphone with my current set up ?

2) If my home phone is currently used, can I still make a call via my smartphone without interrupting the home phone session ?

2) where can I get the manual that explain the meaning of code using to configure OBi.

Example: {username>{(Msp1):sp1},{101>7771234567:aa},{ph}.

In the above code, I want to understand what {(Msp1):sp1} instrucs OBi to do ? what {101>7771234567:aa} instructs OBi to do? etc. etc.

Thanks

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azrobert
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« Reply #35 on: March 06, 2014, 07:34:24 am »

Quote
You means make a call using CSipSimple dialer?

No, you would still use the native dialer.

Quote
Is this where I set the speed dial for the OBi: Users Settings>Speed dials>

Yes

Quote
1) how do I send a text message from my smartphone with my current set up ?

Without a cell phone plan I don't know.
You can login to your GoogleVooice account and send text messages there.

Quote
2) If my home phone is currently used, can I still make a call via my smartphone without interrupting the home phone session ?

Yes. GoogleVoice supports 2 concurrent calls.

Quote
3) where can I get the manual that explain the meaning of code using to configure OBi.

The manual really doesn't explain things very well.

I don't have time now. I'm going to a dance class with wife in one hour and I have to get ready. I'll explain when I get back.

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azrobert
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« Reply #36 on: March 06, 2014, 11:39:44 am »

I posted the explanation of the InboundCallRoute here:
http://www.obitalk.com/forum/index.php?topic=7405.0
« Last Edit: March 09, 2014, 08:58:36 am by azrobert » Logged
HH235
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« Reply #37 on: April 19, 2014, 02:37:23 am »

Really want to thank you for posting this.  It has made my day. 
FYI for some reason I could never get the attendant to route my incoming 101 Android Callcentrtic extension to the aa.  Then i noticed that somehow the "Authorized User Name" on the Obi 100 was 17771234567100. 
So i thought I will change it to just 17771234567.
Then it worked!
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HH235
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« Reply #38 on: April 24, 2014, 09:02:13 pm »

OK I give.  Anyone know how you call another Callcentric # using a traditional phone connected to the Obi 100?
I'm not talking about using a service provider, I  mean IP Freedom # to IP Freedom #.  Like how you can dial another Obi device by just dialing that nine digit Obi #.

Thanks.
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azrobert
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« Reply #39 on: April 24, 2014, 09:27:08 pm »

If it is 2 separate IP Freedom accounts dial the account number of the 2nd account.

If it is one IP Freedom account with 2 extensions dial the extension number (like 101).

You would need the proper routing setup in the OBi to route the number to Callcentric.

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