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