[VoIP] All-new Hack !

[Greg Blakely]

Any changes for Asterisk 1.4?  The patch applied well, and zaptel
compiled, but asterisk barfed.

 

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
> On Behalf Of ikjtel
> Sent: Saturday, August 20, 2005 8:18 AM
> To: Voice Over IP Tandem for Analog Switches
> Subject: [VoIP] All-new Hack !
> 
> As folks suspected, asterisk silences the audio and doesn't 
> cut-thru until after dialing is completed. 
> This behavior is unacceptable to some demented souls...
> 
> This new hack is for anyone in CNET.  When an inward call is 
> placed to your vintage switching office, the caller can now 
> hear the entire call-setup sequence including 
> siezure/offhook, dialtone, and dial pulsing.
> 
> Also, I have added support for 2600 Hz inward SF pulsing.  If 
> the caller transmits a 2600 SF tone, the FXO card will send 
> an onhook signal toward the vintage switching machine for the 
> duration of the tone.  When the tone is removed, the FXO card 
> will go back offhook.
> 
> Provided that WAN packet loss is low enough*, real-time SF 
> dial pulsing should be possible.  That is, the caller (who 
> must have SF outpulsing equipment) can direct-dial through 
> your switch, and hear the individual line finders, selectors, 
> and connectors operate (assuming an SxS office). The caller 
> can also disconnect without breaking the VOIP connection.
>      *- (Also may not work with compressing codecs, etc).
> 
> I would like to ask for beta-testers to help test this patch. 
>  It  requires a recompile of asterisk and zaptel, but your 
> configuration files don't need to be touched.  Also, this is 
> completely unrelated to the x100p "fxs adaptation" hack.
> 
> For more information, please go to
> http://www.lightlink.com/mhp/sf/sf.txt
> 
> Max
> 
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