[Asterisk-video] how to setup SIP-324M gateway

[Arnold P. Siboro]

I got my Asterisk box running and tested with ISDN line. Furthermore, h324m_loopback()
also worked perfectly. I want to setup a SIP-324M gateway, following the
instruction on libh324m gateway, I set it as follows:

[from-zaptel]
exten => _.,1,Answer
;exten => s,10,h324m_gw(SIP/1002)
exten => _X.,1,h324m_gw(s at threegvideo)

[threegvideo]
exten => s,1,Dial(SIP/1002)

However, it does not work (caller keeps ringing but callee does not get
does not response), giving the following message:


Connected to Asterisk 1.4.7.1-BRIstuffed-0.4.0-test2 currently running on asterisk1 (pid = 22790)
ionerisk1*CLI>
Verbosity is at least 3
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
    -- Going to extension s|1 because of immediate=yes
    -- Accepting voice call from '0948523078' to 's' on channel 0/1, span 1
    -- Executing [s at from-zaptel:1] Answer("Zap/1-1", "") in new stack
  == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
    -- Executing [h at from-zaptel:1] Answer("Zap/1-1", "") in new stack
    -- Hungup 'Zap/1-1'

Is it codec problem? I was kind of expecting some message if it's about
codec. BTW, on the SIP end I am using X-Lite, which I think does not
have the right audio codec to talk with h324m endpoint.


Arnold P. Siboro (asiboro at maltech.jp)

"Imagination is more important than knowledge." 
                                 --Albert Einstein.