Third party sip device CUCM - Dialing problem

[denverelway]

Hi,
We have a Bose ControlSpace EX conferencing processors with a 2-line VoIP configured as a Third party sip device on our CUCM 12.5. So far, the device has been add to CUCM (registered with no problem) and we are able to  dial number inside the network (4 digits) and outbound local numbers (9 + area code +  number). The problem occurs when i want to dial long distance numbers (9 + 1 + area code +  numbers). We don't get the tone from our long distance service provider to enter our long distance code and the call is going nowhere. The SIP device is configured exactly as a normal Cisco phone with the same Calling search space.  I did DNA and it is routing on the correct end device (PRI) as I would expect it to.  That part looks normal. On the Bose web interface there is not much configurations. Any suggestions on this problem?

[Gregory Brunn]

Your route patterns are what are enforcing the FAC correct? FAC = force authorization code. Last I checked this was not supported for third party sip devices.

[Gregory Brunn]

If that is the case have you tried creating new route patterns that do not need the FAC. Use a separate CSS to hit these patterns. Basically you would lose the ability to limit who could call long distance via FAC, but you may get the phone working.

[Gregory Brunn]

If the carrier is what is enforce the long distance code, what message is played before you have to enter them.  I have not seen a carrier limit long distance via a code you have to enter. I have seen PIC codes that need to be entered to select the long distance carrier, before but from my memory these need to be dialed before. 

Also I typically have seen carriers like Verizon limit who can call long distance called on the ANI number. If that is the case, make sure your external phone number mask in CUCM is one that is allowed to dial long distance.

[denverelway]

On my route pattern the "require forced authorization code" option is not checked. I think it's configred on the lines provider side

[Gregory Brunn]

Sounds that way, what is your connectivity to your carrier like? PRI, SIP ETC?

If the carrier provides this, is it a DTMF relay problem?

What happens when you call into a toll free number from one of those phones, one that has an option to select something or like a webex bridge, do those DTMF tones actually work?

[denverelway]

My connectivity to the carrier is PRI.

I try a stadard tool free number and it is working so I've made a CDR reporting and when I'm dialing a long distance call (9 + 1 + area code + number) in the report in the Dest No partion  and called No partition fields I get a null value. It seems like it doesn't hit the right pattern of the CSS

[Gregory Brunn]

So your DTMF tones are 100 percent working with a 1800 number.  

Is your external phone number set correctly on the third party sip device?

Can you capture a debug isdn q931 on the voice gateway it egress for long distance and post that along with a standard phone that is working output.

That would be how you can confirm your problem is your not sending a ANI outbound and or your not properly matching what you think you are.