Tek-Tips is the largest IT community on the Internet today!

Members share and learn making Tek-Tips Forums the best source of peer-reviewed technical information on the Internet!

  • Congratulations SkipVought on being selected by the Tek-Tips community for having the most helpful posts in the forums last week. Way to Go!

HELP Callmanager integration

Status
Not open for further replies.

acollard83

IS-IT--Management
May 1, 2005
179
0
0
US
I am using Cisco CallManager for our PBX. We are going to be using Asterisk for our conference server and to "host" some SIP accounts as well as DIDx for our users DID's. I have Trixbox installed and I have the conferencing working great. What I am having trouble with is calls to the CUCM from the Trixbox. I have a DID from DIDx setup, it is going to the trixbox and terminating fine there, I need it to go to the CUCM. I have been scrounging around the internet for any documentation but all is for te base Asterisk, it doesn't work with trixbox/freepbx. I need to get this working. I know very little of Asterisk, thats why I went with trixbox. My forte is Cisco. Any help would be appreciated, a walkthrough/step by step would be awsome.
 
That's exactly what I was looking at. It just deosn't help me with Trixbox. I have no place to put a destination in for a sip trunk did to go to an extension on the Call Manager. We have no extensions, nor do we want extensions on Asterisk.
 
For ease, just put in a catch-all destination. Nothing in the field above and the destination is your conference extension. That way any call coming in over the trunk is sent to the conference service.
 
Go to inbound routes, leave all field blank, Got to set destination at the bottom and choose your conference extension. Save. Apply.
 
We have the calls flowing properly from the CM to Trixbox, it's Trixbox to CM we need to get working.
 
So, if you have no extensions on Asterisk, and do not want extensions on Asterisk, what device is going to initiate a call to get the Call Manager?
 
No device. Just SIP trunks to some service providers, incoming calls from DIDx.
 
So if an call comes in from a SIP provider to Asterisk, set the destination to an Queue, example 2000. In Queue 2000 put an extension on CallMgr. Set an outbound route so that if someone dials the Cisco extension, it routes out over the CallMgr SIP trunk to terminate on the Cisco.
 
Status
Not open for further replies.

Part and Inventory Search

Sponsor

Back
Top