Tzah 2.0


Posts Tagged ‘SIP’

How to make incoming calls from SIP client to OCS 2007 R2

Sunday, May 31st, 2009

Lately, I’ve been experimenting with Office Communication Server (OCS) 2007 R2. I’m checking voice routing between OCS and PSTN. It took me some time to enable incoming calls from a SIP client to reach an OCS communicator so I though I’ll share it with others who are facing the same issue.

First of all, you’d need a SIP client. Normally I use X-Lite but since OCS requires TCP as an underlying protocol to transfer SIP, I used the latest 4.0 version of X-Lite. After installation, setup your account; click on the Transport tab and choose TCP as the signaling transport.

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Now let’s go to the front end server of OCS 2007 R2. Go to the users list (Forest, Enterprise pools, choose the relevant pool and click on ‘Users’ directory), double-click on a user name and select “Configure…” for “Telephony Settings:”. Enter a unique phone number in “Line URI:” field with the format tel:+<country code><Area code><number>. Press OK and you’re good to go!

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Alrighty then. Go back to your SIP client and type the number (don’t forget the plus sign). Your communicator will ring. answer it. Yey.

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Skype is opening-up for businesses. Well, it should open up more.

Wednesday, March 25th, 2009

Skype has made a huge move this week announcing Skype for business. Organizations that have IP-PBX systems can connect it to Skype using SIP. The service, which is currently in beta, has two major benefits to businesses. First, it allows Skype users to connect to the organization using their account. The second, it enables the organizations to make PSTN calls using SkypeOut and saves money.

This is a great day for the VoIP industry. Once Skype users decide to use their account to make a call to an organization, rather than make a regular phone call, the trend towards an all-IP networks would gain more momentum. Furthermore, coupling Skype for business with the use of twitter for direct contact with customers, this service demonstrate the strength that the web/IP has to offer for organizations.

Another reason to celebrate is that Skype, that uses its own standards for signaling and media, has finally integrates with the SIP standard which make them more interoperable. However, it seems that for now, the integration has limited the service functionality to mere calls. That means no presence information, file sharing, text chatting and other great features which only VoIP (we should think of a new name) can deliver. If this remains true, people might start wondering why they should invest their time and energy with a new service that provides them the same functionality they had so far. I truly hope that Skype will enhance the service or otherwise, this entire experiment might cause damage to our industry.

Finally, there’s another aspect that we in the VoIP industry should ponder about. Individuals, as well as organizations, have abundant ways to get in touch with. We have PSTN numbers, emails, twitter, Skype, Windows Messenger, Yahoo messenger, GoogleTalk, and the list goes on and on. What we need is one URI which will map to the specific communication channels of the user. Another service is a global golden page for all those channels. Let’s say I’m looking for John Smith. the service will provide me its URI which in turn, upon sending IM or calling it, will map the request to John’s favorite channel. There’s much more to discuss about this so I’ll leave it to a future post.

Jajah is the VoIP player that brought you web-activated telephony.