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Using as a free PSTN-to-SIP gateway for your FreeSWITCH provides  free access to their DID gateways around Russia and rest of the world. After registration, you get a 5-digit user ID and a password to register on their SIP gateways.

In incoming calls the original Caller ID is lost, so we get some garbage in the From: header, and our own user ID in To:.

Add a new file (conf/sip_profiles/external/zadarma.xml) with your SIP account details:

  <gateway name="zadarma">
    <param name="username" value="XXXXX"/>
    <param name="password" value="XXXXXXXXX"/>
    <param name="extension" value="attendant"/>
    <param name="expire-seconds" value="120"/>
    <param name="register" value="true"/>
    <param name="register-transport" value="udp"/>
    <param name="proxy" value=""/>
    <param name="retry-seconds" value="30"/>
    <param name="caller-id-in-from" value="false"/>
    <param name="ping" value="25"/>

Here “attendant” is a valid extension in my public profile. It directs the call to the automatic attendant IVR menu.  The “extension” directive in the SIP profile sets the value as the destination number.

The attendant extension in my public context (conf/dialplan/public/50_users.xml) looks like the following. It transfers the call to the extension 7800 in my default context:

  <extension name="pub_attendant">
    <condition field="destination_number" expression="^attendant$">
      <action application="transfer" data="7800 XML default"/>
.... (rest of the public profile skipped)

As a result, anyone from around the world is able to access my PBX at local rates.

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Free DID resources

If you want to be easily reachable from abroad, there is a choice of various DID providers, and some of them offer free service.

SIP Broker is a public service where DID providers share their facilities and use a common database for calling SIP destinations. Unfortunately the original company that started SIP Broker is out of business, and it’s unclear who is maintaining the service now. My emails to their support address remain unanswered for the last couple of weeks. But the service works quite fine. and provide lists of DID providers worldwide, including the free ones.

Zadarma is a Russian VoIP provider, and they offer free DID numbers throughout Russia and also worldwide.

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New age telephony is here, yet undervalued

While working on some telephony projects recently, I was pretty much surprised how most people are unaware of the revolutionary changes in the technology in the past decade.

Many (if not most) people, including IT engineers, still think that IP Telephony is about saving on long-distance calls. Some of them also know that it’s about saving on desktop cabling.

Actually it’s much more than that.

1. Low entry cost for office telephony.

With the rise of good-quality, open-source telephony products, such as Asterisk, FreeSWITCH, and a bunch of UI tools on top of them, a small or medium business is not any more facing the need to spend $10000+ on an office telephony system.

An IP PBX can be built in-house, with commodity hardware and open-source software components. Even better, one can rent a hosted PBX if the broadband connection permits that. There are also lots of solutions  for non-IT businesses on the market.

2. Global team infrastructure.

It is now quite common even for small businesses to have remote employees across half the globe. Also the local telecommunication infrastructure is not always readily available.  In some countries, international calls are still ridiculously expensive.

Remote workers can use IP phones to connect directly to your IP PBX. But sometimes the Internet connection at home is not good enough, or your team member is not always at their work place. There are still good and low-cost ways to connect.

For example, DIDWW can offer you a PSTN gateway in remote country, and place the incoming calls directly to your IP PBX.  The SIP Broker allows the same service free of charge, with some extra digits to dial to reach you from their gateways.

3. Voice applications.

Most people still know only two or three voice applications: the voice mail, the conference bridges, and the interactive voice menus. Tor those three, the adoption cost has dropped dramatically. Now one can build any of them at the cost of hardware and an engineer’s hourly work.

But there’s more than that.

One can build voice applications which were too expensive or impossible with traditional PBXes.

For example, employees can choose themselves how they are reachable in the best way. They can manage their own reachability and presence. Calls can be sent to multiple IP and PSTN phones at the same time.  On-call support engineers can login themselves and indicate that hotline calls should be directed to them.  Voice mail can be connected to a ticketing system and create new tickets with MP3 attachments.

those are the applications which I have personally implemented. There are certainly more.

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Virtual hotline: web page

Now going official:


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Virtual hotline service: almost ready

The PBX configuration is finished, with all the call routing as desired.

The SMS gateway service is ordered from  iNetWorx.

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Virtual hotline service: work in progress

status update on the project: Hotline routing works as described below. Yet to be done: self-service menu for the engineers.

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Setting up a virtual hotline service

I’m building a support infrastructure for a small team of engineers (, Data and Voice Operations). Here I will describe some useful design notes, tips, and challenges.

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