In asterisk ONE:

*CLI> dial 302
    -- Executing Dial("OSS/dsp", "SIP/302@81.190.162.42") in new stack
    -- Called 302@81.190.162.42
    -- SIP/81.190.162.42-466e is making progress passing it to OSS/dsp
    -- SIP/81.190.162.42-466e answered OSS/dsp
 << Console call has been answered >>
  == Console is full duplex
  




In asterisk TWO:
Extension 302 calls meetme app 


    -- Playing 'beep' (language 'en')
    -- x=0, open writing:  /var/spool/asterisk/meetme/meetme-username-7777-1 format: sln, 0x82ebe20


Now in asterisk TWO I wrote a call file with:
Channel: SIP/62.xxxx.xxx.xxx-082ca0d
Context: trunklocal
extension: 34657897123



An I get this:

Oct  3 19:42:33 WARNING[14313]: chan_sip.c:1989 create_addr: No such host: 62.xxx.xxx.xxx-0822baf0
Oct  3 19:42:33 NOTICE[14313]: channel.c:2491 __ast_request_and_dial: Unable to request channel SIP/62.xxx.xxx.xxx-0822baf0
Oct  3 19:42:33 NOTICE[14313]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 0