Hi Ram, Its a VOIP Switch based on SIP Proxy. Its Voice Master from SysMaster.
VoiceMaster only authenticates IP and cant have username password based authentication which asterisk can do. So i need to take some traffic from VoiceMaster to Asterisk and terminate it. Let me know Thanks Dan On 01/01/07, ram <[EMAIL PROTECTED]> wrote:
On 12/31/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hi Carlos, > > Im interested in knowing how we can connect 2 server using SIP. Well for > me both are not asterisk servers, 1 is asterisk and 2nd is an SIP based > Server. i need to take multiple calls from the SIP based server and > terminate it using my asterisk peers based on my dialplan. I can use SIP > only as my other server doesnot support IAX2. > > How can i get that. Please let me know Hi what does it mean, sip based server ? please do mention what is that server, most of the servers are SIP based only. ram _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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