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Forward DID to Obihai

Started by Chao, September 27, 2011, 03:58:44 PM

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Chao

I have config my Obihai device as this way:

Service Provider 1: GV
Service Provider 2: Nonoh (which I use to make international calls)

And I have a international DID which I want to map it to my Obihai device. I can not map it to username.nonoh.net since Nonoh does not take incoming calls. So my question is:

1) May I setup the 3rd service provider so that I can add another SP which is able to take incoming calls then I will map my DID to that?

2) I used to have Asterisk box in my home. I was able to map my DID to my Asterisk. I understand Obihai has "kind of" Asterisk service embedded. Is there anyway I can map my DID to Obihai device in "Asterisk way"?

Thanks.

RonR

Can your DID be forwarded via SIP URI?  If so, simply forward it to:

sip:anything@hostname_or_ipaddress:5061

where hostname_or_ipaddress is your public hostname or IP address and the port number (5061) is for SP2 (must be confiured for SIP).  For SP1, use port 5060, but SP1 would have to be configured for SIP.

If your DID requires SIP registraion, you would have to put it on SP1 or SP2 as only two SIP registrations are supported.

RonR

I forgot to mention...

If you use forwarding via SIP URI, you will need to forward the SIP port through your router to your OBi.

Chao

Thanks for your prompt reply. Forwarding sip url does make my phone ringing, but after I pick it up, there is no sound in either way.

I forwarded port 5061 and 10000-20000 to Obihai device on my Tomato router. Is there anything I need to do? Thanks.

RonR

Unfortunately, there are a number of people experiencing audio problems with the OBi, but there is no response from Obihai on the issue.

Everton

#5
Since you only use the Nonoh account for outgoing, maybe what you want to do is setup Nonoh on Voice Gateway 3 (IF it doesn't require registration with their Sever to place outgoing calls) and setup another SIP account on SP2.  Then forward your calls to your OBi device via SIP URI to that provider, which will handle the NAT issues.  In my case, I have a free UK DID from UKDDI and forward all incoming calls to my VOIP.ms account (which is configure on SP2) via SIP URI.  Maybe a free Callcentric IP Freedom account will serve this purpose if Nonoh doesn't accept inbound SIP URI calls.

Also, you could try to setup your OBi/Router for direct inbound SIP URI calls by following this generalize setup for the Linksys device (Note:  There are some Linksys features that is note readily apparent on the OBi device, so maybe OBiSupport could provide some help in this regard):
---------------------------------------------------------------------------------------------------------
Here's a more general set of steps to properly set up a PAP2/SPAxxxx type ATA behind a NAT router, that should work under just about any conditions.

Browse to the ATA's advanced, admin level web configuration. Under the SIP tab:

1) Record the port number specified by RTP Port Min.
2) Record the port number specified by RTP Port Max.
3) Set Handle VIA received to yes.
4) Set Handle VIA rport to yes.
5) Set Insert VIA received to yes.
6) Set Insert VIA rport to yes.
7) Set Substitute VIA Addr to yes.
8 ) Set Resp to Src Port to yes.
9) Set STUN Enable to yes.
10) Set STUN Server to your choice for STUN server, eg "stun.voxalot.com".

11) Set Ext RTP Port Min to match the value recorded in step 1). May be omitted depending on router's NAT implementation.

Under the Line 1/2 tab:

12) Set NAT Mapping Enable to yes.
13) Optionally, set NAT Keep Alive Enable to yes. Isn't really necessary with ports forwarded.
14) Make a note of the value of field SIP Port.
15) Set Ext SIP Port to the value from step 14). May be omitted depending on router's NAT implementation.
16) Set Ans Call Without Reg to yes.

In your router's setup:

17) Forward the port number from step 14) for UDP, from your router to the ATA.
18) Forward the range of port numbers defined by the values from steps 1) and 2) for UDP, from your router to the ATA.

Finally, in your "international DID provider's" setup, make sure that the SIP forwarding goes to the same port number as recorded in step 14).

Chao

#6
Thanks for your detailed reply. I would like to try the gateway method first.

http://www.obitalk.com/forum/index.php?topic=1063.0