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Silicon Valley Products, Corp.: Linking Servers using SIP

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  Linking Servers using SIP Original AMP enhancements

The original utilities to enhance ('Enhanceme') AMP, A@H, Asterisk and Cisco XML. and Speed Dialing are now in the download area (They still remain on Sorceforge as well). This download also includes the original Paging using custom meetme routines. Intercom by dialing 0 + Ext #. browsing AMP's extensions on your phones LCD and the Backup patch for Asterisk@Home maint screen.


The paging and Cisco phone display reoutines are still appropiate. The speedial routine has been converted to module form for FreePBX, but it is not in this old AMP release.  I will document the new speeddial module and put it in the upload area in the near future.

Paul Norris

  
 
  Posted by master on Friday, December 15 @ 17:02:29 EST (1714 reads)
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  Linking Servers using SIP Using SIP instead of IAX to Link Servers


I like to have links with my client’s machines, but I found no real information on how to do this by SIP.  There are a lot of scraps everywhere, but Asterisk to Asterisk is constantly referenced and described using an IAX link.  If you think about various suppliers like Stanaphone (one of my favorites) linking to their server with yours is documented.  This is a SIP server to SIP server, but where on VOIPinfo is the configuration for both sides?


If such a configuration is documented, I missed it.  So I decided to rediscover the wheel again and document it.  I supply Asterisk systems based on the AMP and now FreePBX control programs so my configurations are based on these modules.

Lets continue onto the meat...

  
 
  Posted by master on Saturday, November 25 @ 14:13:44 EST (7787 reads)
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