[Asterisk-Users] Callmanager 4.1 and Asterisk

[Gonzalo Gasca Meza]

You need to create a SIP trunk in CCM and in Asterisk a peer in sip.conf with the IP address of the CCM (trunk)
In the trunk configuration change the transport to UDP.
Enter the IP of Asterisk.
And create a route pattern with gateway the SIP trunk
 
In Asterisk in extensions.conf create the route to CCM phones.
I have this setup in my lab with CCM 4.02sr1 and works so fine.
If you need the sip.conf / extensions.conf and an screenshot of the route pattern and SIP trunk config just let me know!
Happy holidays!
 

Keith O'Brien <keitheobrien at yahoo.com> wrote:

I have a similar setup.   To make it easy and get the best of both worlds, have the Linux softphones (SIP or IAX) register to Asterisk.   Keep the physical phones registered to CM.   From there setup a dialplan on both Call Manager and Asterisk to relay calls between the two systems.   For example, assign all physical phones extension 2XXX and softphones 3XXX.   Have asterisk route 2XXX calls to CM via SIP and vice versa on Call Manager.

Also, just so that you are aware you can register a SIP Linux softclient to Cisco Call Manager if you are running Version 4.1

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Hello everybody,

im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager.

We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux machines, i read about callmanager uses skinny a propetary protocol and there are no softphones from linux to talk with it, so we need to install vmware to use ipcommunicator or the other solutions as i read is get the asterisk server using sip phones in the linux and windows machines and configure the call manager to talk with the asterisk server thru sip protocol, is this the real way to do that?? is there a easy way to do this?? i found this link

http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration

but i need to know what things to do to transfer all the extensions from de callmanager to the asterisk sw, or if only made the changes in the sip.conf as said in the link above the callmanager gets all the control??

or if i need to declare all the extensions in the asterisk?? can anybody help me??

TIA

Edgar




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