Basic SIP phone as OBi110 client

[Hagrid]

I've searched this topic in both the manual and forums for a while, but all I can find are complex (but interesting!) setups that go far beyond what I want.  I just need some help with the basics.

I have an OBi110, one SIP provider, one analog phone, and one Cisco 7941G SIP phone.

I want to be able to use both the analog phone and SIP phone to place calls to the SIP provider by default. Incoming calls should ring on both sets. I also want 911 to go to the line port.  (There's no POTS service, but I live in an area where the telco must still maintain 911 service on disconnected lines.)

So far, I've set up the service provider and I can make and receive calls with the analog phone. (Took 2 minutes - awesome!) The SIP phone is set not to register, connects to the OBi, and I get dialtone, but as soon as I dial the first digit the analog phone rings.

Can anybody point me at the next step?

[azrobert]

I'm assuming you are pointing your SIP phone to SP1 on the OBi110.
The call from the SIP phone is coming into the OBi110 as an inbound call on SP1 and ringing the Phone Port. You need to change the configuration to bridge the inbound call out SP1 or Line. For inbound calls from your provider you need to fork the call to the Phone Port and the SIP phone.

Service Provider -> SP1 Service

X_InboundCallRoute:
{cisco>(Msp1):sp1},{cisco>911:li},{ph,sp1(cisco@192.168.1.100:5060;ua=$1)}

MaxSessions: 4

cisco = UserID used in the SIP phone setup
192.168.1.100:5060 = IP address of the SIP Phone

When you make a call on the SIP phone it uses two channel sessions. One for the call to the OBi110 and another for the outbound call. I changed MaxSessions to allow an inbound call from your provider while using the SIP Phone.

I did not test the above config.

[Hagrid]

> I'm assuming you are pointing your SIP phone to SP1 on the OBi110.

AHA! That would be the case if I left it at the default port 5060, right? And since that is also where my SIP provider is, I can see why I get the loop around.

Your suggestion to bridge that call out SP1 is to leave SP2 free, right?  So alternately I could point the SIP phone to SP2 by using a different port instead?  Not saying I should, just trying to understand the background.

> When you make a call on the SIP phone it uses two channel sessions. One for the call to the OBi110 and another for the outbound call.

So one channel between OBi and phone, and one between OBi and provider?

Thanks a lot.

[azrobert]

Your suggestion to bridge that call out SP1 is to leave SP2 free, right?  So alternately I could point the SIP phone to SP2 by using a different port instead?  Not saying I should, just trying to understand the background.

The call from the IP phone to the OBi is an inbound call. Just how it works.

SP2 is free and you could setup SP2 to receive the call from the IP phone by changing the port number, but you will have to setup SP2's X_InboundCallRoute to bridge the call.

So one channel between OBi and phone, and one between OBi and provider?

yes.


If you haven't made any changes yet, call the OBi again from the SIP phone. Answer the call and see if you have 2 way audio. Sometimes these setups have audio problems.

[azrobert]

I had 2 errors in the config in my 1st post. I changed the portion in error to the following:
{ph,sp1(cisco@192.168.1.100:5060)}

I used SP2 to route inbound calls to the IP phone and I forgot name@ in the address.
name can be anything. SP2 has not been defined, so you can't use it to route calls.

[Hagrid]

The call from the IP phone to the OBi is an inbound call. Just how it works.

OK, that makes more sense.  I was thinking of it as an outgoing call.

SP2 is free and you could setup SP2 to receive the call from the IP phone by changing the port number, but you will have to setup SP2's X_InboundCallRoute to bridge the call.

So an "SP" is a service profile then?

If you haven't made any changes yet, call the OBi again from the SIP phone. Answer the call and see if you have 2 way audio. Sometimes these setups have audio problems.

I did.  Made talking to myself way more fun than normal. 

[Hagrid]

I had 2 errors in the config in my 1st post. I changed the portion in error to the following:
{ph,sp1(cisco@192.168.1.100:5060}

I used SP2 to route inbound calls to the IP phone and I forgot name@ in the address.
name can be anything. SP2 has not been defined, so you can't use it to route calls.

Shouldn't there also be a closing parenthesis before the closing brace?

{cisco>(Msp1):sp1},{cisco>911:li},{ph,sp1(cisco@192.168.1.100:5060)}

Calls from the SIP phone to anything matching the sp1 digitmap route to sp1, from SIP phone to 911 routes to line out.  All other calls are sent to both the analog phone and the cisco phone using sp1.

Gold star?

[azrobert]

Shouldn't there also be a closing parenthesis before the closing brace?

{cisco>(Msp1):sp1},{cisco>911:li},{ph,sp1(cisco@192.168.1.100:5060)}

yes. I'm having a bad day with the finger checks.

Calls from the SIP phone to anything matching the sp1 digitmap route to sp1, from SIP phone to 911 routes to line out.  All other calls are sent to both the analog phone and the cisco phone using sp1.

Gold star?

Yep. You understand 100%

[Hagrid]

Thanks a lot for the help.