[asterisk-users] Native bridging analog phones trouble DAHDI channels.

Kevin P. Fleming kpfleming at digium.com
Tue Sep 29 07:23:07 CDT 2009


Maurizio Faccio adinet wrote:
> I own a TDM2400 board, with three FXO modules and one FXS.
> I'am having trouble with analog sip phones, from two different 
> equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), 
> sometimes when I am calling someone, then I press flash, and then call 
> someone else, both calls stay connected after I hang up.

That's because you have just completed a flash-hook based transfer of
the first call to the second call. If you don't want this feature, set
'transfer=no' for the relevant channels in chan_dahdi.conf.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org



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