Skip to main content

Dinstar GSM gateway FreeSwitch HowTo

About​

Dinstar's products include E1/T1 gateway, VoIP GSM/CDMA gateway, access gateway, softswitch, IP phone, billing systems, and Internet voice value-added service total solutions, etc., which are characterized by complete specifications and good compatibility. Moreover, Dinstar VoIP products reaches an operation level truly in respect of availability, manageability, and maintainability.

Click here to expand Table of Contents

Dinstar GSM Gateway HowTo​

This is the minimal basic configuration to make the Dinstar GSM Gateway work with FreeSwitch.

Dinstar GSM Gateway HowTo ASSUMPTIONS​

You have the default Dinstar GSM Gateway configuration.

You have the initial FreeSwitch install.

You have are familiar with FreeSwitch dialplan and CLI commands.

Dinstar Device Before you Start​

Remember to remove the PIN number from your SIM card. Double check your SIM card with a Phone. It should not ask you for a PIN. Check out that the led lights (green and red) are blinking as they should.

Dinstar Device Login to FS​

Dinstar main menu -> System Configuration -> SIP Configuration -> add SIP Server Address(freeswitch IP:172.16.33.100) and Is Register=yes

System configuration->port configuration -> select port 0, add user id:1000, Authenticate ID:1000, Authenticate Password:1234, add to VoIP hotline:1001

Operation->Tel->IP Operation->add "any", tick the auto call

This will redirect all your inbound calls to this Freeswitch extension (1001) in this case

  • Sip proxy should aim to the IP where your FS is running.
  • Do not change any other parameter, they should be as they are by default.
  • Make sure same all settings and go to tools->reboot to reboot the system

Dinstar Device Status​

System inforamtion -> System Information, Mobile Information, SIP Information

  • Make sure the system is on with working status.
  • Make sure SIM cards have been detected correctly and show card infomation.
  • Make sure SIP is regiestered.

FreeSwitch Gateway Configuration Example​

Edit the /usr/local/freeswitch/conf/sip_profiles/external/example.xml

<include>
  <gateway name="gsm">
  <!--/// account username *required* ///-->
  <param name="username" value="1000"/>
  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
  <param name="realm" value="172.16.33.10"/>
  <!--/// username to use in from: *optional* same as  username, if blank ///-->
  <param name="from-user" value="1000"/>
  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
  <!--<param name="from-domain" value="asterlink.com"/>-->
  <!--/// account password *required* ///-->
  <param name="password" value="1234"/>
  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
  <!--<param name="extension" value="cluecon"/>-->
  <!--/// proxy host: *optional* same as realm, if blank ///-->
  <!--<param name="proxy" value="asterlink.com"/>-->
  <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
  <!--<param name="register-proxy" value="mysbc.com"/>-->
  <!--/// expire in seconds: *optional* 3600, if blank ///-->
  <!--<param name="expire-seconds" value="60"/>-->
  <!--/// do not register ///-->
  <param name="register" value="false"/>
  <!-- which transport to use for register -->
  <!--<param name="register-transport" value="udp"/>-->
  <!--How many seconds before a retry when a failure or timeout occurs -->
  <!--<param name="retry-seconds" value="30"/>-->
  <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
  <!--<param name="caller-id-in-from" value="false"/>-->
  <!--extra sip params to send in the contact-->
  <!--<param name="contact-params" value="tport=tcp"/>-->
  <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
  <!--<param name="ping" value="25"/>-->
  </gateway>
</include>

FreeSwitch Context (Outbound) Configuration Example​

  <extension name="GSM inbound">
    <condition field="destination_number" expression="^(1000)$">
      <action application="log" data="INFO do you inbound stuff here!!!!! ani --${ani} "/>
      <action application="hangup" />   // can transfer to any exten(10001,1002...)
   </condition>
  </extension>
  • In this example all the inbound calls are sent to the extension 1000 then they are hanged up. The ani should appear.
  • 1001 also would ring, if the call is coming. please make sure 1001 is regiestered in freeswitch.

Troubleshooting and Useful Tips​

  • Press F4 under fs_cli, make sure it is there:
   external::gsm       gateway                      sip:1000@172.16.33.10      NOREG


  • sofia status (to check that the Dinstar gateway is properly configured and it is running).

freeswitch@internal> sofia status gateway example.com

Name example.com Profile external Scheme Digest Realm example.com Username joeuser Password yes From <sip:joeuser@example.com;transport=udp> Contact <sip:gw+example.com@172.16.33.100:5080;transport=udp;gw=example.com> Exten 5000 To sip:joeuser@example.com Proxy sip:example.com Context public Expires 600 Freq 600 Ping 0 PingFreq 0 PingState 0/0/0 State NOREG Status UP CallsIN 0 CallsOUT 0

  • If you are having trouble with the outbound calls try to log directly into the Dinstar Gateway, that is to say create an account with your favorite SIP client (Jitsy, X-Lite, SipDroid, CSipSimple...) use the same username and password and remember that the IP must aim at your Dinstar Gateway IP, not the Freeswitch one, then when your client is logged, try to make a call.

Useful Docs​

how to set freeswitch and asterisk with Dinstar gateway

See Also​

Dinstar Official Website

Sofia_Configuration_Files#Structure_of_a_Profile

FreeSwitch_Dialplan_XML

Comments:​

鼎信通达 Posted by livem at Jan 22, 2018 06:35