Sip route error

[cwlinde]

I have to problems:

1. The time conditions on my inbound sip routing doesn't work.
2. At Call Routing, when I choose "Route all incoming SIP calls to Call Routing" I get a message "The number you have dialed is unavailable", when dailing the number. When I untick it, phoning the number works perfectly.

I have attached a screen shot of my inbound route. I route everything through extention 999.

Hope someone can help

[davrays]

Well, Ill try to help:

1. How do you know they don't work? They used to work everywhere else. More details, please... :)
2. When you select the "Route all incoming SIP calls to Call Routing", the calls are routed directly into CRT, without looking for the destrination in the "Extensions Management Table". So you should have a rule which will match the destination of the incoming call. Can you show the INVITE message which comes from the server. What do you see in the "TO" header (the reouting decision is made based on that header)?

The rule which you have attached, will match only if the destination of the call is exactly 0875500867. I doubt that is the case. And even if this rule will match, your call will be routed to the PBX extension 9990875500867. I am almost sure there is no such extension on your device, and it is natural that the incoming call gives "The number you have dialed is unavailable".
I guess this rule is incorrect and should be modified. If you tell what you wanted to achieve with this rule, I may be able to help you..

Best regards,
David