Connecting Yeastar TG200 GSM gateway to FreeSWITCH

I needed to connect a GSM gateway to my FreeSWITCH PBX, in order to receive SMS and mobile calls and emulate a normal mobile phone. I’ve got the Yeastar Neogate TG200 V2 for this purpose (Firmware Version: 53.18.0.39, running Asterisk 1.6.2.6 on an ARM processor).

This blog entry has helped a lot and saved a bunch of time.

The box supports OpenVPN, so you can place it in some remote location behind NAT, and manage it via the VPN connection. The client version is rather old (2.0.5), so it does not support embedded certificates in the client config, and also “topology subnet” option is not supported. You need to pack your vpn.conf and the certificates and pivate key into a TAR archive and upload to Neogate via its web interface.

It’s sufficient to configure one SIP trunk to your PBX, and manipulate the To: header in order to distinguish between SIM cards on incoming calls.

When the SIP trunk was configured (FreeSWITCH as a registrar), I started receiving the following warnings on FreeSWITCH, and the registration status was quickly removed after neogate’s REGISTER message:

2014-12-22 12:29:42.208567 [WARNING] sofia.c:5721 Sip user 'gsm01@xxxxx.net' is now Unreachable
2014-12-22 12:29:42.208567 [WARNING] sofia.c:5732 Expire sip user 'gsm01@xxxxx.net' due to options failure

My FreeSWITCH was sending SIP OPTIONS requests to all registered users and removed the registrations unless the clients responded with status 200 or 468. Neogate responds with 404 Not Found on such requests toward the trunk SIP user. I had to disable “unregister-on-options-fail” option in FreeSWITCH internal SIP profile.

In SIP trunk configuration, “Advanced->Caller ID” was automatically set to my trunk’s registration user name. Because of this, all incoming calls had this name as the caller ID, and the original caller number was lost. After setting this field to blank, the problem was resolved.

In “Mobile to IP” rules, you can set a different rule for each SIM card. The “Hotline” field should not be blank, and should contain some distinguishing number. It will be used in To field in the SIP INVITE on incoming calls. If you leave “Hotline” empty, the Neogate will respond with dial tone and collect DTMF digits before placing the call to your SIP trunk. So far I could not find any documentation that describes this.

Also in trunk configuration, sometimes I had to reboot the box in order for my changes to take effect.

The box uses the standard Asterisk management interface for sending and receiving SMS, and I’m planning to use this Perl module through the VPN connection.

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