i apreciatte if u can send me the conf files, and the screenshots about the CM config, its really easy as you said, i like asterisk very much, after that we are planning to make test on echo and relay calls, but think it would work great, thanx for your help,
Edgar > > 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 <[EMAIL PROTECTED]> 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 As terisk 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 > > ----------------------------------------------- > > 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 > > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > --------------------------------- > Do you Yahoo!? > Yahoo! Mail - Easier than ever with enhanced search. Learn > more._______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
