Coding OBI for 1 to many SIP accounts per SP account

[Hortoristic]

Sure would be nice if I could enable one of my SP's to handle many SIP accounts.  Such as 1 to many usersnames and passwords.

I've got DID numbers with various providers around the world (mostly getting the free ones) - but each instance eats away at a SP.

Something like this:

AuthUserName: GVUserName; voipMSUserName; IPKallUserName
AuthPassword: GVPassword; voipMSPassword; IPKallPassword

[N7AS]

I have several DIDs coming ito one SIP account. For example, I have several GV numbers routed to several Callcentric Free NY DIDs. I then forward these DIDs from Callcentric to one Anveo account via SIP URI setup on my OBi110. I also have some UKddi DIDs forwarded via SIP URI to Anveo.

[Hortoristic]

I think I just found it - an article RonR wrote; here is his quote - I think this means I can have 1 to many DID's all pointing to a SP that is configured for SIP - I'm not getting his workaround for Caller ID routing tips either:

If you have one or more DID's that forward via SIP URI to an ATA, softphone, or hardphone, they can be forwarded to your OBi instead.

Your OBi must have SP1 and/or SP2 configured for SIP.  It need not be a working provider.  You can set the ITSPx proxy server to anything (like 127.0.0.1) and disable X_RegisterEnable on SPx.

Forward ports 5060 and 5061 in your router to your OBi's IP address.

Forward your DID to your public IP address and the appropriate port number (SP1 = 5060. SP2 = 5061).  If you don't have a static IP address, a Dynamic DNS host name works fine.  Use any phone number:

12345@xxx.dyndns.org:5061

Calls will arrive at the PHONE Port, but CallerID can used on the SPx -> X_InboundCallRoute to perform the normal routing tricks.

I'm using this with IPComms and IPKall and both work very reliably.

[rsriram22]

I have a question though - would the CNAM/caller ID get forwarded for inbound calls to obi's SIP URI (say from callcentric)?

i guess i am going to try this out. i want to setup the following:

1) SP1: GV (without a GV number) to be used for outgoing calls, by default *AND* incoming callcentric calls via SIP URI.

2) SP2: Anveo E911 (@80 cents/month, its the cheapest E911 around)

wow, feeling a bit richer again (although by few cents/month - $1.50/month E911 on callcentric to anveo)..

It sounds theoritically possible, so let me give it a spin!

Thanks @hortoristic pointing out RonR's thread regarding SIP URI ports. I almost started a new thread regarding the mapping of ports 5060/61 to SP1/2..

[jimates]

Once you set up GV on SP1 that is all you can use on that trunk.

Without a google voice number, your outgoing calls from SP1 will carry a random caller id from google and if anyone tries to call back using that number will get a failed call.

IMO, the extra 70 cents per month for CC vs Anveo is well worth the simplicity of use.

[rsriram22]

(after some more reading on the forums that only SIP enabled SPx can be used for SIP URI usage, i am now thinking that i should be able to use SP2 to make E911 *and* receive inbound callcentric calls on SP2 (via SIP URI forwarding) - no?

[jimates]

I can't say yes or no as I am not familiar with the requirements of Anveo or sip uri calling; or how they will/can interact with CC.