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« Reader Question: How do I text a message to somebody's cell phone from my computer? Does VoIP allow this to happen? | Main | Pain Point - Demystifying the VoIP Business Value »

Reader Question: We have been doing project on asterisk PBX. we configured it well and it is in testng phone using SIP softphone. Now we have a problem connecting a HARD phone to the asterisk server. Please suggest some useful information on connecting

Here's another reader question that came in:

We have been doing project on asterisk PBX. We configured it well and it is in testing phone using SIP softphone. Now we have a problem connecting a HARD phone to the asterisk server.  Please suggest some useful information on connecting the hard phone to the ASTERISK PBX.

Now I'll confess that I'm not at all knowledgeable about Asterisk in depth. It isn't a fit in the VoIP environment I most frequently work in. The only dabbling I've done with Asterisk is in my home office and I only use softphones at present. So I resorted to Google and found this among the datasheets at VoIPSupply.com.

I'm not going to cut and paste the entire chapter referenced, but it's only three pages long and looked to me like it had enough information. You might want to find the entire document this was excerpted from. it looks reasonably helpful. And here's some really helpful Asterisk tips from the beginning of the document I found.

Much of the information in the book came from the Asterisk Wiki pages at http://www.voip-info.org/tiki-index.php. This is a gold mine of Asterisk information. While I have mined some of the gold, there is still a lot left for you to find. The Asterisk community done a tremendous service to the community in creating this resource.

You can get help from the Asterisk mailing list. Consult www.asterisk.org for more information on support and the mailing lists.

Digium, of course, offers Asterisk support and free support for issues related to their hardware. Signate, www.signate.com, is in the business of supporting Asterisk.

This is posted on both the blog and in the Realtime VoIP Community forum, so readers can also freely jump in to help answer any specific questions.

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The Team has reAsterisk Development leased version 1.2.7.1 of Asterisk.This release versions is available on the Digium FTP servers as PGP signed tarballs and also as PGP signed patch files, so that it could be easily upgraded from the previous versions of asterik.

Discuss voip related topics at http://www.voip-traffic.com

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Ken Camp's Bio:

Ken Camp has more than 25 years of experience in information technology. Ken spent 17 years with AT&T and Lucent Technologies successfully designing and implementing voice and data networks. He later worked in the security marketplace and played a key role in early IPSec VPN deployments. As an independent consultant, Ken's primary focal areas include network performance improvement, security practices and the design and deployment of integrated voice and data solutions. He may be contacted at: ken_camp@realtimepublishers.net

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