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No number showing on VoIP Phone

Started by Orien2k, March 14, 2017, 05:13:22 AM

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Orien2k

Hello, i'm trying to configure a Obi110 and a Asterisk FreePBX in a RaspberryPi. I can make calls to the world and receive calls from the world. But when i receive calls from outside all numbers come as OBITRUNK1 that is the login of my SIP. Is there a way to forward the number that is calling me so that it shows up on my 3CX Softphone ? I forgot to mention the Obi110 is configured as FXO Gateway my phone line is connected to the Obi110. The other persons gets our number on their phone.

azrobert

#1
Turn on callerid spoofing on the trunk to FreePBX.

Service Providers -> ITSP Profile x -> SIP -> X_SpoofCallerID: Checked

Orien2k

It's turned On. on the FreePBX logs i can't find my number when i call the Obi110

drgeoff

On which incoming calls is this failure to display the number happening?

1. POTS calls on the LINE input of the OBi110.

2. VoIP calls coming in on an Asterisk trunk.

Orien2k

Hello, the problem occurs when I call from my Mobile phone to our Phone Line lets say my phone line is 22 555 10 20 and i call this number from my mobile..
I call my phone line and in my PC on the Softphone software (3CX) the number shows OBITRUNK1 (this is the Trunk Login). If i call an extension it shows the person name and the right extension.

On the Obi110 i have no Phone connected all calls are forward to the 3CX apps on the PCs.

azrobert

The following is how I setup RasPBX and an OBi110. I designed this method by myself. I modeled it the way 2 OBi devices can talk to each other. This is different than other tutorials I've seen. They have the OBi110 register as an extension. I don't and route inbound Line calls to the IP address of Asterisk. This requires you to define a trunk on FreePBX with the IP address of the OBi110. Without this trunk Asterisk will treat the call as Anonymous and reject it.  I dumped RasPBX because GoogleVoice stopped working. I'm now using Incredible PBX on my Raspberry Pi. I only have a GoogleVoice trunk defined, so I'm not setup to test and see if this method still works with a newer version of FreePBX/Asterisk.

Asterisk:
Setup an outbound trunk without registration for the OBi110:
Peer Details:
type=peer
username=OBi110
host=192.168.1.110       (IP Addr of OBi110)
port=5060                        (Port# of SP1)
canreinvite=no
insecure=invite,port
qualify=yes

OBi110:
SP1 must be defined as an SIP trunk to send/receive to/from Asterisk.
Setup a dummy SIP definition like this:

Service Providers -> ITSP Profile A -> SIP -> ProxyServer: 127.0.0.1
Service Providers -> ITSP Profile A -> SIP -> X_SpoofCallerID: Checked
Voice Services -> SP1 Service -> AuthUserName: OBi110
Voice Services -> SP1 Service -> X_RegisterEnable: (unchecked)

Route outbound calls from Asterisk to Line
Voice Services -> SP1 Service -> X_InboundCallRoute: li

Route inbound Line calls to Asterisk
Physical Interfaces ->Line Port -> InboundCallRoute:
sp1(900@192.168.1.100:5060)  (Asterisk IP address and port)
900 is an extension or you can setup an inbound route for 900