[asterisk-users] call from h323 to SIP

[nik600]

On 12/15/06, Pavel Jezek <pavel.jezek at i.cz> wrote:
> probably you haven't g729 installed in asterisk, use g711 instead, put
> this in h323.conf and in callmanager place asterisdk gateway in region
> that will use g711...
> disallow=all
> allow=alaw
>
> alternatively you can find g729 codecs binaries here:
> http://kvin.lv/pub/Linux/Asterisk/
>
>

I am experiencig the same problem:

h323.conf:
disallow=all
allow=all               ; turns on all installed codecs

sip.conf:
disallow=all                    ; First disallow all codecs
;allow=all                      ; Allow codecs in order of preference
allow=g711
allow=ulaw

extension.conf:
exten = 3298,1,Set(SIP_CODEC=alaw)
exten = 3298,2,Answer
exten = 3298,3,Dial(SIP/user at 193.x.x.x)

    -- Executing Set("H323/ip$172.z.z.z:1630/20", "SIP_CODEC=alaw") in new stack
    -- Executing Answer("H323/ip$172.z.z.z:1630/20", "") in new stack
    -- Executing Dial("H323/ip$172.1z.z.z:1630/20",
"SIP/user at 193.x.x.x") in new stack
    -- Called user at 193.x.x.x
    -- SIP/193.x.x.x-40451408 is ringing
    -- Got SIP response 606 "Not Acceptable" back from 193.x.x.x
  == No one is available to answer at this time (1:0/0/0)
  == Auto fallthrough, channel 'H323/ip$172.z.z.z:1630/20' status is 'NOANSWER'

What means 606 ?