Unable to do blind transfer

an trying to configure blind transfer and atended transfer in Asterisk ,my configuration files are here below. lets say
caller A calls to B , when B press *2 for transfer the log file says rtp codec unknown.



features.conf
[general]
parkext => 700
parkpos => 701-704

context => parkedcalls
parkingtime => 45

transferdigittimeout => 3
courtesytone = beep

;xfersound = beep
;xferfailsound = beeperr
;adsipark = yes
;findslot => next

pickupexten = *8
;featuredigittimeout = 500

[featuremap]
blindxfer => #1
disconnect => *0
automon => *1
atxfer => *2

Extensions.conf
exten => 401,1,Dial(SIP/401||tT)
exten => 402,1,Dial(SIP/402||tT)
exten => 403,1,Dial(SIP/403||tT)
exten => 404,1,Dial(SIP/404||tT)
[applicationmap]
; Note that the DYNAMIC_FEATURES channel variable must be set to use the features
; defined here. The value of DYNAMIC_FEATURES should be the names of the features
; to allow the channel to use separated by '#'. For example:
; Set(DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3)
;
;testfeature => #9,callee,Playback,tt-monkeys ;Play tt-monkeys to
;callee if #9 was pressed


Any ideas?
HELP

Hello.
I think that you have wrong configured your DTMF relay.
Check if in channel config (SIP/H.323/IAX2) and at client side the same methods of DTMF transmission.
If you use RFC2833 DTMF relay you should use 101 payload type for such relay - not 96 by default.

Its an educated guess, but the 'unkown codec' probably refers to the fact that one of your UA's only supports codec's that Asterisk doe not know about.

If you capture the output of 'sip debug' at the asterisk CLI when you test your blind transfer, you should be able to see which is the offending UA.

Cheers,
X-C

Thanks a lot
I recompiled asterisk and the rtp unkown codec error is gone
now ....

I get error message as bridge failed on channels

I called B from A
In B pressed *2 i am not getting dial tone and the call disconnects and trows this error message

res_features.c:1384 ast_bridge_call: Bridge failed on channels SIP/401-08bb7e20 and SIP/403-08bbaa40

here I am trying to transfer teh call to 402 extension

plz help
thanks in advance

Thanks a lot
recompilation of asterisk worked

my issue resolved