The VoIP Lounge

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Incessant ramblings of a Unified Communications enthusiast with sporadic moments of sensibility.

About this blog

The idea behind this blog is to publish news, events and advancements in the world of IP Telephony and voip to share technical documentation as well as my personal experiences while working on deployments and general troubleshooting. I will also occasionally share design documentation that should assist those readers interested in pursuing IP Telephony and VoIP design.

4 Responses

  1. Hi Asad,

    I look forward to reading more. I enjoy your writing style and think that you will make a very interesting blog.

    Best of luck,
    Daren.

    Like

  2. Asad Raza says:

    Hi Daren. Welcome to my blog, and thanks for your feedback. Due to my hectic schedule, I haven’t had the time to post anything (other than the welcome message). Your post has served as a reminder that I need to resume (or start) blogging. Thanks for visiting!

    Like

  3. Robert Showerman says:

    Advertising Rates On asadraza.wordpress.com?

    Hi,

    I was wondering how much you would charge to advertise on your site. I am looking to place a small text based add. If the price is affordable I may buy more than one over time. I usually ask for one line with just a few words. I prefer them at the top, middle, or bottom of the main content area of the site. However, I am sure we can work out placement.
    Please let me know how much would you charge to place a link ad on your site? If you are interested let me know the price and if you prefer PayPal or any other mode of payment?


    Thanks & Regards

    Robert Showerman

    Marketing Manager
    Robert.showerman@gmail.com

    Like

  4. Irina says:

    Hi Asad!

    We’re a Cisco Solution Partner specializing in software development for Cisco Collaboration and Contact Center solutions. We’d also like to publish some article on your blog asadraza.wordpress.com.

    Please contact me.

    Like

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Did you know?

The SIP protocol does not carry the number 'type' information for calling number such as 'international' or 'subscriber' etc. Therefore, for incoming calls at the PSTN, the calling number needs to be manipulated at the gateway before the call is routed to the call agent over a SIP trunk.