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I want to make my obi110 POTS line accesible over a SIP URI

Started by chuby007, June 09, 2012, 08:26:46 AM

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chuby007

Hi I have a   :o :o :o ***BRAND NEW****  :o :o :o ( lol )

no really I have a new  obi 110, I want to allow acces to my POTS line via a sip URI

I want to dial **8-Number@ip:port but I have no idea how to do it...

Right now I have a GV on SIP1 but I can easily delete that...

I just want that SIP URI tu work

THX !

pc44

Hi chuby007,

I'm not sure I understand exactly what you are trying to do.  Are you wanting to receive incoming calls via SIP URI?  Or, are you trying to dial (outgoing) a specific SIP URI from your phone?

If you could confirm your goals here, perhaps someone can help.  In the meantime, here are a couple threads that *may* be helpful to you:

And this one is if you are interested in using the Jitsi softphone with your OBI for SIP URI dialing:
Need help automating sip uri for "Wakeup calls"

pc44 :)

chuby007

what I want to make is this:


        Sip URI ---- > OBI ----> POTS

Basically I want to dial my pots over a sip uri.

I do have google voice set up, but that SIP URI---> OBI ----> POTS is the only thing that interests me... so I can delete the google voice account if required !

pc44

Hi chuby007,

Hope I am now correctly grasping what you are trying to do.  First try these instructions from the threads I linked to earlier.  It sounds like your SP2 is currently unused; so go ahead and use SP2 for this setup if it's available.

Quote from: RonR on November 05, 2011, 09:31:51 AM
In order to receive incoming calls via SIP URI, you must have at least one service (SP1 and/or SP2) configured as SIP.  It need not have an actual SIP provider operational on it, but by being configured as SIP, it cannot be used for Google Voice.

Your unused SP2 can be configured for SIP as follows:

Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1

Voice Services -> SP2 Service -> AuthUserName : (put anything here)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B

SP2 status should show : Registration Not Required

You will also need to forward port 5061 to your OBi.

SIP URI's can then be addressed to : anything@obi100_public_IP:5061

Now, you will need to forward all inbound calls to your Line Port, if I am understanding you correctly.  You do want to forward ALL inbound calls to your POTS, correct?

Quote from: RonR on December 14, 2011, 07:31:06 PM
If you're wanting all incoming callers to your Google Voice number to be able place calls to specific PSTN numbers, set:

Voice Services -> SP1 Service -> X_InboundCallRoute : {aa}

Voice Services -> Auto Attendant -> PrimaryLine : PSTN Line

This will route all incoming callers on your Google Voice number to the Auto Attendant where they can press:

1 to continue the call to the OBi PHONE Port
2 to make a new outgoing call (using the LINE Port by default)
3 to have the Auto Attendant make a callback

You may also wish to set:

Voice Services -> Auto Attendant -> PIN1

to limit access to your Auto Attendant to authorized callers only.

The same settings can be configured for SP2 if you want it to function similarly.

Where the instructions say SP1 above, just use SP2 instead if you are using SP2.  Follow these instructions first to make sure it works successfully.

If the above works successfully and you then want to eliminate the auto attendant, I'm thinking you would do a Voice Services -> SP2 Service -> X_InboundCallRoute : li   If that fails, you may be able to get away with setting an unconditional call forward to {li} by Voice Services -> SP2 Service -> CallForwardUnconditionalEnable -> Checked and Voice Services -> SP2 Service -> CallForwardUnconditionalNumber -> li

I cannot test this out right now or else I would do so for you.  So perhaps you could give it a try and report back with your findings.  If using SP1 instead of SP2, some of the instructions might differ (i.e. Port 5060 instead of 5061).

pc44

Stewart

Assuming that you have GV on SP1, set up SP2 for SIP.  This could be a real SIP provider; many have useful services even on free accounts, or you could make a dummy entry, e.g.:
ProxyServer: 127.0.0.1
X_RegisterEnable: (unchecked)
AuthUserName: 123
AuthPassword: 123

If it's a dummy entry, you can simply set
X_InboundCallRoute: li

Otherwise, you'll need to distinguish calls sent to the URI from calls coming from the provider.

Then, send a call for the POTS line to number@ip:5061
(or whatever port you have set in X_UserAgentPort).

Note that if the port in question is exposed to the Internet, anyone could use your POTS line to call anywhere.  The OBi doesn't have any way to provide real security.  However, you could do one or more of the following:

1. Use X_AccessList to limit the IP addresses than can connect.
2. Set X_InboundCallRoute to send only calls from a "magic" caller ID to the Line port.
3. Set X_InboundCallRoute to accept only calls to numbers with a "magic" prefix (and strip that off).
4. Set the digit map to allow only calls to countries you need to call.

chuby007

Guys I appreciate your help...

I did this:

ProxyServer: 127.0.0.1
X_RegisterEnable: (unchecked)
AuthUserName: 123
AuthPassword: 123

If it's a dummy entry, you can simply set
X_InboundCallRoute: li

Otherwise, you'll need to distinguish calls sent to the URI from calls coming from the provider.

Then, send a call for the POTS line to number@ip:5061
(or whatever port you have set in X_UserAgentPort).

I am able to call out, the phone rings however when I pickup there's no audio in or out...

as I said google voice is not important for me... I just need to make

SIPURL----> OBI ----> POTS

basically have my pots line as an FXO to make my POTS line accesible via SIP URI
it works... however I dial out.. no voice on any side !

2012-06-09_1807 by Techno Kitty Development, on Flickr

pc44

Quote from: chuby007 on June 09, 2012, 05:10:20 PMI am able to call out, the phone rings however when I pickup there's no audio in or out...

as I said google voice is not important for me... I just need to make

SIPURL----> OBI ----> POTS

basically have my pots line as an FXO to make my POTS line accesible via SIP URI
it works... however I dial out.. no voice on any side !

Glad to hear you have it working!  When you say you "dial out" what you are trying to dial?  I thought you only wanted incoming:  SIPURI --> OBI --> POTS?  Isn't that incoming only to your POTS?

pc44

Stewart

Quote from: chuby007 on June 09, 2012, 05:10:20 PMbasically have my pots line as an FXO to make my POTS line accesible via SIP URI
it works... however I dial out.. no voice on any side !
If the calling device is on the same LAN as the OBi, I don't know why it's not working; use SIP Debug to see where RTP is going and where the OBi told the client to send it.

Otherwise, this is likely a NAT issue.  Assuming worst case (both client and OBi are behind separate NATs), try the following:
For ITSP Profile B -> General, turn on STUNEnable, set STUNServer to a working STUN server such as stun.counterpath.net and turn on X_SymmetricRTPEnable.  Also, in your router, forward the UDP ports corresponding to the range specified by LocalPortMin to LocalPortMax to the private static (or statically assigned by DHCP) address of the OBi.  If your router does not support port range forwarding, you can safely change LocalPortMax to 16806 and just forward UDP 16800, 16802, 16804 and 16806.

If it still doesn't work, try setting up STUN on the client and forwarding its RTP port range.

If no luck, post details about your setup (router makes/models, modem makes/models, client device or software).

chuby007

ok it worked like a charm there's a bit of lag but I guess that's normal since I am on WIFI, maybe a better voice compression codec or something ( no idea if I can change it with the obi )

however this is the deal I am calling with my android phone and dialing out using my very cheap landline, so I make all my calls using the landline... not the cellphnoe line however I dial using the cellphone.

It's an android phone and I see lots of people asking how to do it.. . I will post a guide :D

pc44

chuby007,

Dialing out from your cellphone... I see now what you mean.  Have you had any success with the OBiON for Android App?

pc44

chuby007

Obion does work however I wanted for the phone to ask me before I dialed... if I wanted to use my regular dialer or SIP phone... by using CipSimple :D and that worked !