I would suggest you go through your dialplan entries and compare them to what you had before. You might also think of putting a "catchall" rule in for troubleshooting and turn the proxy log up to debug and see what is being sent to sipx in the way of digits.
I saw a "weird" thing too trying to return calls through our Ingate, which is also an unmanaged gateway. I used to have an entry "%2B" that would have to be stripped on returning calls from call-logs on the phone where they showed "+" (and an eleven digit number). In 4.0 that call needs to now send "+" instead of the SIP HEX value of of + as "%2B" in order for it to make it past the dial rule and hit the proxy and get sent. At that point if you tail the log: tail -f /var/log/sipxpbx/sipXproxy.log you might be able to see what the proxy might be complaining about. So In my case I changed the rule from a prefix of %2B to + and now it all works again. Dialing 1 (and 11 digits) always worked, but when the callerid said +1 and you wanted to hit the dial button on the polycom it did not until I made the above change. So I am convinced some items did change in the proxy for sure. >>> Mark Wood <mark.w...@redphonetech.com> 05/05/09 12:54 PM >>> Fresh ISO install of 4.0 Setup an Asterisk box as an unmanaged gateway for interim purposes, also setup associated dial plan. I can make calls from the SipXecs system to asterisk and use the Asterisk box as a PSTN gateway as well. But calls from the Asterisk to the SipXecs do not go through, it (Asterisk) responds with an "all circuits are busy" announcement. If I check the CDR records in the SipXecs system it shows the call with all the correct info but with a failed in the status column. This exact configuration worked in our 3.10.2 system, any additional configs for the 4,0? M Wood _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
