Can not make a call from the analogue phone to the IP one

[beckzayi]

I am using 3cx version 6, and patton 4118.

Here is the scenario. We have a 3CX server on a computer, a Patton smartnode 4118, an IP phone, a normal phone, and the network connecting these components together. We just want to make a connection from the normal phone through Patton to the IP phone.

I did try to import the configuration file (the patton distribuotor sent to me), and that is the only file I imported. The result is that I can use the IP phone to call the normal phone, and also I can make calls between 2 analogue phones, but I can’t use the normal phone to call the IP one (busy sound). Do I miss anything?

Thanks ...

 

[avraam_mich]

You need to set up the Patton 4114 with 3CX according to the instructions found on http://www.3cx.com/voip-gateways/patton-smartnode.html.

In the last step of these instructions, you get an option to generate a custom configuration file which can be uploaded to the device. Please use this file to upload to he Patton.

 

[beckzayi]

Hi there. Thanks for your reply.
I did try what you recommended, and I also tried to 'Add a VOIP provider'. It all worked. BUT, this is not the scenario that I was trying to ask in the opening question.

My question is: I can not use an analogue phone to call an IP phone via the Patton box and 3CX server, no PSTN or VOIP provider gets involoved. The problem is similar to http://www.3cx.com/forums/internal-dial-error-unidentified-incoming-call-5326.html

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16:29:17.328 Call::Terminate [CM503008]: Call(12): Call is terminated
16:29:17.328 CallCtrl:nIncomingCall [CM502001]: Source info: From: 12; To: [sip:[email protected]:5060];tag=484de90b6d12c41[sip:[email protected]]
16:29:17.328 CallCtrl:nIncomingCall [CM503013]: Call(12): Incoming call rejected, caller is unknown; msg=SipReq: INVITE [email protected] tid=3ec4469a8 cseq=INVITE [email protected]:5060 / 567296415 from(wire)
16:29:16.625 evt::CheckIfAuthIsRequired::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;branch=z9hG4bK7a0cf4f5d
Max-Forwards: 70
Route: [sip:192.168.1.156:5060;lr]
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: [sip:[email protected]:5060];tag=484de90b6d12c41
Call-ID: [email protected]
CSeq: 567296414 INVITE
Session-Expires: 1800
Supported: timer, replaces
User-Agent: Patton SN4118 4JS4JO EUI MxSF v3.2.8.45 00A0BA0206A6 R4.2 2008-05-15 H323 SIP FXS FXO
Content-Length: 0
16:29:16.625 evt::CheckIfAuthIsRequired::not_handled [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected] tid=7a0cf4f5d cseq=INVITE [email protected]:5060 / 567296414 from(wire)
---------

Extenstion 101 is the IP phone, 112 is the analogue. 112 can't call 101 due to the authorization issue which I have no idea. But I can use 101 to call 112.
Any idea? Thanks...