News from Industry

Connecting Yeastar TG200 GSM gateway to FreeSWITCH

TXLAB - Mon, 12/22/2014 - 14:53

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:, running Asterisk 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 '' is now Unreachable 2014-12-22 12:29:42.208567 [WARNING] sofia.c:5732 Expire sip user '' 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.

Filed under: Networking Tagged: freeswitch, GSM, pbx, voip


2600hz - Thu, 11/20/2014 - 02:34


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2600hz - Mon, 11/17/2014 - 09:22

Inspiration projects for next-generation Torrus

TXLAB - Sat, 11/15/2014 - 23:46

It’s not yet clear when I can start working on a new-generation Torrus, but here are some nice software projects which would probably inspire the new design, or probably be part of the new design. I haven’t looked into them in depth though.

  • Bosun is a distributed monitoring system produced by StackExchange. It uses distributed collector agents which write data into OpenTSDB. Bosun and its collector are written in Go, and OpenTSDB is written in Java.
  • InfluxDB is a time-series database written in Go.

and yes, the new project will most probably have its core in Go. But the SNMP discovery engine will most probably remain in Perl because of a big list of supported vendors.

Filed under: Programming Tagged: monitoring, network management, network monitoring, torrus

CEO Mike Leuthner discusses Phonami’s new Admin panel,...

2600hz - Sat, 11/15/2014 - 04:09

CEO Mike Leuthner discusses Phonami’s new Admin panel, allowing customers to activate Monster PBX accounts, setup billing, add and remove users directly from Google Apps. 

SendHub co-founder Ryan Pfeffer discusses a product overview,...

2600hz - Sat, 11/15/2014 - 04:02

SendHub co-founder Ryan Pfeffer discusses a product overview, Kazoo at scale, and supporting WebRTC and Mobile.

2600hz and Voxbone discussing WebRTC at WeWork San Francisco,...

2600hz - Fri, 11/14/2014 - 20:09

2600hz and Voxbone discussing WebRTC at WeWork San Francisco, ….sorry for the poor audio quality

2600hz launches new hosted PBX partner website!

2600hz - Fri, 10/31/2014 - 22:20

Want to become your own carrier!  We’ve launched an incredible new website dedicated to partners, which can be found at The new site provides information, resources and contact information for VoIP partners interested in Hosted PBX, and soon, Hosted PBX + Mobile.

What does this mean for you? You’ll find new information about our product offerings and how to market immediately and scale quickly.

On our new partner website, you’ll find resources on:

  • Hosted PBX – Providing small and medium businesses a fixed-line telecom infrastructure without heavy financial investment. There is no bulky hardware to install or maintain, instead the infrastructure is in the cloud.
  • Coming Soon - Hosted PBX + Mobile – We’ve integrated our mobile services with Sprint, providing a native integration with mobile devices.  Our platform will not function as an over-the-top application, instead transforming a cellular phone into a SIP endpoint.

In conjunction with this website launch, we’re introducing a whole bunch of new and exciting features and benefits that include:

  • Advanced Functionality - Conferencing, call recording, automatic failover, find-me-follow-me, hunt groups, voicemail-to-email and more. For a complete list of features, visit
  • White labeling – We provide clients the ability to whitelabel our product as their own and go to market. We also provide white-label documentation including logos, email templates, pricing and restrictions.
  • Bring Your Own Device - We enable you to port customer’s existing mobile devices to 2600hz. BYOD gives you the flexibility to fit a customer’s budget by utilizing existing devices.
  • Flexible Contracts - We give you the ability to structure fixed PBX, and soon mobile pricing as you see fit. You can offer customers monthly or yearly contracts and set margins with bundled packages.
  • Kazoo Integration - Developed by 2600hz, Kazoo employs a cloud-based model designed for cutting edge telecom infrastructures. Kazoo can be deployed in a private cloud environment or can be utilized in a hosted cloud environment.
  • Integration - Kazoo is built on over 140 APIs that allow you to integrate with carriers, third-party software vendors, and build your own complex integrations.

Become a 2600hz Partner Today!

Want to take the first step? Contact or sign up today at We will give you in-depth training and support you as you build your business. Become a valued partner and own the competition!

Also read the press release:

Using Voxbeam for outbound calls with FreeSWITCH

TXLAB - Tue, 10/21/2014 - 18:53

Voxbeam is providing worldwide PSTN connectivity at competitive rates, and it allows you to use any Caller ID, which is very convenient for call forwarding. The Voxbeam gateway authenticates the clients by their IP addresses only, so you need a static IP address, and no username or password are required. The FreeSWITCH configuration shown below allows you to control which destinations should be routed to Voxbeam. With a bit of further extension, you can also control which destinations would use different tariff plans at Voxbeam. This configuration covers only their Standard pricing plan. Here INTERNALDOMAIN is a name of the SIP realm that is used for registered users. We assume that the variable “outbound_caller_id_number” is set elsewhere above in the dialplan.

--- File: ip_profiles/external/voxbeam.xml --- <include>   <gateway name="voxbeam_outbound">     <param name="realm" value="" />     <param name="register" value="false" /> <!-- important, so that your caller ID is transmitted properly -->     <param name="caller-id-in-from" value="true"/>   </gateway> </include> --- File: dialplan/INTERNALDOMAIN/05_pstn_outbound.xml --- <include> <!-- Express destination and caller numbers in E.164 notation without leading plus sign. Note that we treat numbers with one leading zero as local Swiss numbers -->   <extension name="pstn_normalize" continue="true">     <condition field="destination_number" expression="^00([1-9]\d+)$" break="never">       <action inline="true" application="set" data="e164_dest=$1"/>     </condition>     <condition field="destination_number" expression="^0([1-9]\d+)$" break="never">       <action inline="true" application="set" data="e164_dest=41$1"/>     </condition>     <condition field="${outbound_caller_id_number}" expression="^00([1-9]\d+)$" break="never">       <action inline="true" application="set" data="e164_cid=$1"/>     </condition>     <condition field="${outbound_caller_id_number}" expression="^0([1-9]\d+)$" break="never">       <action inline="true" application="set" data="e164_cid=41$1"/>     </condition>   </extension> <!-- Here we define that calls to Russia and Ukraine should go through Voxbeam -->   <extension name="pstn_select_itsp" continue="true">     <condition field="${e164_dest}" expression="^(7|38)" break="on-true">       <action inline="true" application="set" data="outbound_itsp=voxbeam"/>     </condition>   </extension>   <!-- send matched calls to Voxbeam -->   <extension name="pstn_voxbeam">     <condition field="${outbound_itsp}" expression="^voxbeam$" break="on-false">       <action application="set" data="effective_caller_id_number=${e164_cid}"/>       <action application="bridge" data="sofia/gateway/voxbeam_outbound/0011103${e164_dest}"/>     </condition>   </extension> <!-- send everything else to -->   <extension name="pstn_sipcall">     <condition field="destination_number" expression="^(0\d+)$">       <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>       <action application="bridge" data="sofia/gateway/sipcall/$1"/>     </condition>   </extension> </include>

This is a very simple example, and a bit more logic can be introduced, such as looking up in some kind of a database for least cost routing, and so on.


Filed under: Networking Tagged: freeswitch, pbx, sip, voip


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