[Asterisk-Users] phantom answer

[support at sigmasdi.com]

People,
My goal is to get asterisk dialing out via my landline (POTS) from a sip
softphone. Ive got the phone, The TDM400p is installed and working. (See
below) When ever I dial a number that is directed to the outgoing port on my
card (fxs/fxo?) I get no ringing, then it claims its been answered. the CLI
reports the following:
 
 
Executing Dial("SIP/301-f97a", "Zap/4/01614299100|20") in new stack
    -- Called 4/01614299100
    -- Zap/4-1 answered SIP/301-f97a
Jun 15 17:57:38 NOTICE[11121]: rtp.c:277 process_rfc3389: Comfort noise
support incomplete in Asterisk (RFC 3389).  Please turn off on client if
possible. Client IP: 192.168.0.7
    -- Hungup 'Zap/4-1'
 
Anyone Any Ideas? BTW Apologies for the disclaimer at the bottom, but the
mail server adds it on by default and there's nothing I can do about it.
 
*CLI> zap show channels
   Chan Extension  Context         Language   MusicOnHold
 pseudo            default
      1            default                    default
      4            incoming                   default
*CLI>
 
This is the important bit from zapata.conf

; DYLAN ADDED FROM DIGIUM.COM ********************************
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=yes ; Asterisk trains to the beginning of the call, number is
in milliseconds
callerid=01614830073
signalling=fxo_ks
group=1
context=default ; Points to the default context of your extensions.conf
channel => 1
 
signalling=fxs_ks
;callerid=asreceived
group=2
context=incoming
channel=> 4
; END OF DYLAN ADDED FROM DIGIUM.COM *************************
 
 



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