Cisco CallManager 6 to SIP Server via SIP Trunk

1. Brekeke Product Name and version:Brekeke SIP Server & 2.3.6.0/286

2. Java version:1.6.0_14

3. OS type and the version:Windows 2003 & 5.2

4. UA (phone), gateway or other hardware/software involved:X-Lite Softphone & Cisco Unified Communications Manager v6.1

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem: I am trying to set up a Brekeke SIP Server to communicate via a SIP Trunk to a Cisco CUCM v6.1. I am a CCNA, but I haven't had much of an opportunity to work with configuring SIP trunks or route patterns in the past. I work at a hospital and we are implementing a Rauland/Borg SIP enabled nurse call system into a new three story wing. Each patient room will register with the Brekeke SIP Server and then route the call over the SIP trunk to the Call Manager. Our nurses will be carrying Cisco 7921G wireless phones to receive the patient calls. This scenario is currently "un-certified" by Cisco, but they are working with Rauland for a formal solution. We are still waiting for Rauland to complete the hardware installation, but I wanted to get the Brekeke SIP Server communicating with my Call Manager as soon as I can. I've downloaded the free X-Lite soft-phone as a test and I've gotten it registered with the SIP Server as user 6800, but I'm not sure exactly how to configure the dial plan on the Brekeke side or the SIP Route Pattern on the Cisco side. Any help would be greatly appreciated.

i am interested in this as well. Since i am doing the same thing as you Justin. I have Brekeke loaded onto the RGS server (got it working finally by changing the SIP port RGS uses) i have a dial plan created
Matching Patterns:
$request= ^INVITE

Deploy:
$target = IP Address of SIP listener provided by Cisco PBX admin


i don't know if i would be able to call back into the room from the Cisco handset with this dial plan or not. all new to me.

The following DialPlan will work for you.

-------------------------------------
Matching Patterns:
$request= ^INVITE
To = sip:9(.+)@

Deploy Patterns:
To = sip:%1@<Cisco_IP_Address>
-------------------------------------

If you dial with the prefix "9" at SIP client , a call will be forwarded to the CISCO's SIP Trunk.


To make a call from CISCO's side, you need to set the Brekeke SIP Server's IP address at CISCO's settings.

Thanks for the info Laurie. I edited my dial plan like you suggested, but I keep getting Call Not Found when I dial an external number. We do use 9 for our outside access code. I figured out how to successfully make a call to a Cisco 7940 that is registered to my CallManager using the Skinny protocol. I can successfully call any four digit extension in our 8000 range by using the following dial plan:

-------------------------------------
Matching Patterns:
$request=^INVITE
To=sip:(8.+)@

Deploy Patterns:
To=sip:%1@<Cisco_IP_Address>

I was also able to successfully call from a Cisco 7940 back to the soft phone by using this dial plan:

-------------------------------------
Matching Patterns:
$request=^INVITE
To=sip:6800@

Deploy Patterns:
$auth=off
To=sip:6800@

Since the dial plan appears to be correct, I will also be opening a TAC case with Cisco to ensure I have the SIP Trunk, the Partition and Calling Search Spaces set up correctly on the Cisco side.

Im glad to know you got a progress.

Your Dialplan ruels seem no problem.

If you have any update, post it here!

I was able to get this working Justin. I have the Brekeke actually loaded on the Rauland Gateway server, which in turn uses port5060. so i had a conflict there. So i had to change the Responder Gateway Server to use a different port (i used 5061). In addition, i had to set the RTP to "ON" rather than use the "Auto" function. I would get the SIP connection from the Room to the Cisco Handset fine or Cisco Handset to Room, but had no audio. After hours of staring at it, I changed the RTP = "ON" and it worked right away! Also i setup the Nurse Call Registrations with a prefix of "**" and it seemed to work fine as well. So the Dial PLans on the cisco call manager wouldn't be dependent on a specific numeric prefix.

Thanks all for your support!
Brad

Hi Brad,

Thank you for sharing the useful information.
I appreciate it.

Thanks
mitu@brekeke.com

Thank you for the information Brad. I appreciate it very much.

I've been in contact with Cisco and I sent them a few call traces and they determined that the outside access code (9) was being stripped off before it got to the CallManager. Is there anywhere in the SIP Server configuration to modify this? Also, how do the Rauland patient rooms register to the SIP Server? Is there a way to differentiate between buttons? For example, if I wanted all of the patient room buttons to go to a Cisco wireless phone, but I want the Emergency button to go to another Cisco phone that we have set up for emergencies in our ER Registration area?

Thanks,
Justin

Please add '9' before %1

-------------------------------------
Matching Patterns:
$request= ^INVITE
To = sip:9(.+)@

Deploy Patterns:
To = sip:9%1@<Cisco_IP_Address>
-------------------------------------

Justin to direct your calls for the emergency calls in the Rauland system you would have to have "Teams" built in the Responder5 system. For example: Code Blue Team
This team would be set to only receive calls from code blue buttons.
Then in the Responder5 application, the members of that team would have to go "On Duty" for the "Code Blue Team". Then they would receive the call for the Code Blue calls. In addition, the code blue call can notify any staff members on duty for the unit that the code blue is pressed with no additional configuration just by selecting the "Urgent" checkbox in the staff list in the Resonder5 Application. you may want to double check, but i am thinking that the devices can call a maximum 12 devices. Normally it is best to create this as a TAP output to send a text message to the phones if you have the appropriate middleware in place. SOunds complicated, but very doable.