I am having an issue with trying to route all call that come in on one of the FXO lines over a SIP Trunk to my Asterisk box. Here is what I have working and not working:
Exchange 2010 UM (Voice Mail)
CISCO 2621XM with VIC2-FXO ports, connected as Gateway in CUCM
FXO Ports configured on CUCM 6.1 - these are working ona VIC2-FXO card
I have IP Phones configured and connected to CUCM.
I Have a SIP Trunk configured for Asterisk connections.
I have a ROUTE PLAN that says any 5XXXX Number route over the SIP Trunk
I can call the Asterisk Box just fine when I use the IP phones that are connected to CUCM such as EXT 1000 on CUCM can call any extension on the Asterisk Box that are 5XXX.
I can call from the outside and reach an extension on the Asterisk Box as long as the phone is login.
If I setup the FXO port for a DN say 5000 and I dial in and the phone is not login I get a BUSY as I have all of the extensions setup to route to Exchange 2010 voice mail.
I need to figure out how to hard code a CALLER ID number from the CISCO FXO Port as when it tries to dial the Exchange Voice Mail, this is where I am getting a BUSY. These FXO lines from the PSTN do not have CALLER-ID and I beleive this is what is causing this issue with BUSY.
So what I need is advice on how to setup the FXO PORT with a CALLER-ID Number or some type of FIXED Number that is sent to Exchange.
Any help would be appreciated.
first remove mgcp from router: "no mgcp"
then you need 2 things:
1. Configure PLAR for fxo so when incoming call hits port it goes to plar configured, then...
2. Configure dial-peer for the previous plar
connection plar 5000
dial-peer voice 1 voip
session target ipv4:x.x.x.x ----->call manager's ip address
codec ?(depending on region)
note that this affects all your dialplan so if outgoing calls is to be passsed through this it will also need to be changed. Here are some exapmles:
does it help ?
Are you using MGCP? If so, it doesn't support caller-id, and you must use H.323 instead.
station number ...