[hamarin]

Pardon my ignorance but I'm a bit at a loss trying to figure this stuff out. I currently have a working Cisco VOIP infrastructure with Cisco phones, Cisco Voice Gateways, Cisco Call Manager and Cisco Unity. I recently aquired a client who is currently using a software based SIP phone and I am trying to work with them to move to a hardware based SIP phone for fault tolerance reasons and that integrates with my infrastructure. After some research, it looks like the Grandstream HandyTone 286 will be an ideal device for them based upon cost and other considerations imposed by the client. Now that being said, I've been searching the Internet trying to find some guidance on how in the world I integrate SIP phones into my Cisco infrastructure. Can anyone gently provide a pointer to any existing FAQs, whitepapers, whatever that can help me to understand how this works? Thanks in advance.

[mberlant]

I recommend that as your first move you log in to your CCO account and search for "Grandstream". I have no idea if anything is there, but there might be. After that, Googling would be my second step.

You are doing the inverse of what we usually do (we generally integrate Cisco phones with generic clients), but your post might very well attract the help you need. Good luck.

[hamarin]

I did all the searches you talk about, but none of it is very clear to me. I also found and printed out a 300+ page document about SIP implementations from Cisco and have been reading through it. Nearly all of the documentation I'm finding on Cisco's site talk about making their 7940s and 7960s work as SIP phones. I sent an email to my Cisco SE yesterday and got this response: "CallManager 4.0 and 4.1 do not support SIP endpoints, including Cisco products or 3rd party products. This support is coming in CallManager 5.0 in Q1 2006. CallManager does support SIP trunks today. I will try to reach you on Friday to discuss this project live."

I'll post an update if I manage a working solution. Maybe I need to deploy a SipX server that has a trunk to my Call Manager until v 5.0 comes out.

[mberlant]

I was going to recommend Asterisk as your link, just because I am more familiar with it, but SipX should work just as well.