The problem seems to be that, typing 801 on a phone connected to the UC560 the “To” field contains "@192.168.0.230" (which is the IP address of the UC)
Unlike a SIP call made by a PC connected to UC560 with SotPhone attested to Asterisk I note the following:
The Field “To” is properly compiled!!!
There’s a way to Edit the “To:” field of the incominc SIP Call?
Can you give me some guidance to solve the problem?
Asterisk will not normally look at the domain part. In any case, it routes based on the request URI, not the one in the To: header, and the request URI only appears in the INVITE.
You need to provide the contents of the INVITE and all the responses to it, not just the final ACK. Also, just in case you made the routing domain sensitive, you need to provide the contents of sip.conf.
You should probably team up with the person who has incoming CUCM calls working but can’t make outgoing ones (I’m not sure if he posted to the correct forum, so you may want to look at other ones, if you can’t find the posting I’m referring to.
and normally strip out everything except 801, and then use that to look up an extension in the context that was matched based on the source address, and possibly the From line user part.
Basically, though, you are continuing to provide insufficient information to debug the problem. You still haven’t provider the SIP response, which will be telling the other side why the call was rejected. With the right debug and verbosity settings, Asterisk will also be telling you why it is rejecting the call. (Generally, you should uncomment full, in logger.conf, and run with core set verbose 5 and core set debug 5, then provide the logs from /var/log/asterisk/full.)
Also you haven’t provided sip.conf. There are some, rarely used, options in there that can change the context used depending on the domain part of the URI in the INVITE line.
Asterisk does not normally route based on the domain part.