routing all calls thru asterisk 11.2 using Obi202 and obi110 with pstn

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azrobert:
Judging by your question I assume you have an Asterisk setup without FreePBX and you're probably not familiar with FreePBX.

I looked at a sample sip.conf and extensions.conf. It looks like sip.conf is the equivalent of FreePBX's trunk and extension definitions and extensions.conf is the equivalent of the inbound and outbound routes.

Here is my trunk definition:
type=peer
username=robert1
host=192.168.1.101
port=5060
fromuser=robert1
canreinvite=no
insecure=invite,port
qualify=yes
nat=yes

I took the default settings for my extension definitions except for Password and Dial.

I have no idea how this translates to sip.conf.

It looks like the Dial function is in extensions.conf.

cssobi:
Robert,

when I enter a 11 digit number I see it go to asterisk (CLI) but then it rings busy


css

azrobert:
Make the following change.
Obi110
SP1 service X_InboundCallRoute
{robert1>(Msp1):li},{>101:ph},{ph}

This will ring the OBi110 phone port.
This will indicate you're call is getting to the OBi110, but something is not matching.

We are going out for dinner, so I'm signing off for the night.

cssobi:
thanks for your input,
I was able to get the fundamentals working [will be trying with 3 Obi202s soon]....
the reason I was getting a busy was because of an extra parenthesis in extensions.conf ...

on your post with adding a trunk for obi202 i.e. the one with robert2
I haven't added that ... when might that be useful or needed?

azrobert:
I'm not thinking clearly and making things more difficult than it has to be. I have 2 OBi110s and I call the extension on the other OBi110 by passing the extension number out the Asterisk trunk, therefore I need a trunk on Asterisk for each OBi110. I can use the other OBi110's SPx trunks the same way. I had to do it this way because all my OBi110 SPx trunks are used for Service Providers and I can't use them for Asterisk extensions. I use a Voice Gateway to send commands to Asterisk. Anybody out there that thinks I'm crazy to use Asterisk this way, I DON'T. I setup Asterisk as a learning experience and normally have it powered off.

Anyway, it doesn't make sense for you to do it this way since you have the SPx trunks for Asterisk extensions. Here is my revised suggestion for your setup:
 
Asterisk

Setup a trunk for the OBi110:
type=peer
username=robert1
host=192.168.1.101 (OBi110 IP addr)
port=5060

Setup Extensions 101, 201 and 202


OBi110

Register SP1 as Asterisk extension 101.

ITSP A DigitMap:
(1xxxxxxxxxx|xxxxxxx|011xx.)

ITSP A -> SIP
X_SpoofCallerID: Enabled (checked)
This is needed to pass CallerID to Asterisk for inbound calls on Line.

Phone Port DigitMap:
((Msp1)|201|202|911)

Phone PortOutboundCallRoute:
{911:li},{((Msp1)|201|202):sp1}

Line InboundCallRoute:
{sp1(999@192.168.1.100:5060}  (Asterisk IP address)

SP1 service X_InboundCallRoute:
{robert1>((Msp1)|911):li},{ph}

OBi202

Register SP1 as Asterisk extension 201.
Register SP2 as Asterisk extension 202.

ITSP A DigitMap:
(911|1xxxxxxxxxx|xxxxxxx|011xx.)

Phone Port1 DigitMap:
((Msp1)|101|202)

Phone Port1 OutboundCallRoute:
{((Msp1)|101|202):sp1}

SP1 service X_InboundCallRoute:
ph1

ITSP B DigitMap:
(911|1xxxxxxxxxx|xxxxxxx|011xx.)

Phone Port2 DigitMap:
((Msp2)|101|201)

Phone Port2 OutboundCallRoute:
{((Msp2)|101|201):sp2}

SP2 service X_InboundCallRoute:
ph2


Edit:
You should include the default rules for the Phone Port DigitMaps and OutboundCallRoutes.

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