[cisco-voip] URI dialing to VCS (CallManager 9.1)

In CCMAdmin, under CallRouting SIP Route Pattern I don't see where I can send a single URI (appears to be whole domain or ip address or nothing) to VCS. For example from Jabber for Windows 9.6.0 (my number is cool...@coke.commailto:cool...@coke.com I dial torfina...@coke.com, I want that

Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number

thanks all, I have implemented using MVA hairpinning. works well On 12 February 2014 04:38, Buchanan, James jbucha...@presidio.com wrote: I agree with Matthew--I had completely forgotten about MVA. It works reasonably well. James Buchanan | Sr. Network Engineer Presidio | www.presidio.com

Re: [cisco-voip] URI dialing to VCS (CallManager 9.1)

Jason, You set a route string on the URI Catalog such as vcs.coke.com and then just create a SIP Route Pattern to match and point it towards the VCS SIP Trunk. Brian From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Jason Aarons (AM) Sent: Thursday, February 13, 2014

[cisco-voip] SIP Stack Trace Question

Folks: I'm reviewing an issue and looking for some feedback on a specific SIP stack error. In a nutshell, what appears to be happening is CUCM (8.6.2.24090-1) responding to an UPDATE request with a 500 Internal Server Error in a very specific transfer call flow: .::: Call is established :::

Re: [cisco-voip] SIP Stack Trace Question

Brian: Thanks for the response. I'd have to review traces for a non-transfer scenario but it's my understanding that, in this scenario, the 183 w/SDP from CUCM was the answer in the offer/answer model - the offer being Nortel's INVITE w/SDP. Just after allocation of the MTP, SIP Interface

Re: [cisco-voip] SIP Stack Trace Question

Daniel, That seems to be correct from the RFCs I just took a quick look at. Would have to take a look in the SDLs to see if CUCM is actually putting the SIPInterface into a sessionEstablished state after it sends the 183 w/SDP. It may still be waiting for the Prack though before it marks it

[cisco-voip] Calling Party Name and Calling Calling Party Number in a SIP Trunk to VCS

Running CallManager 9.1 with VCS 8.1 Is there a standard on how a SIP Video endpoint should show Calling Party Name and Calling Calling Party Number when using URI dialing in SIP to SIP or SIP to H323 (VCS Internetworking) Can I show a Friendly name like Jason Aarons vs

Re: [cisco-voip] Calling Party Name and Calling Calling Party Number in a SIP Trunk to VCS

hy what about a LUA script on the Sip Trunk, to put the right Display Name the customer wants. Ther u can cut the left side and cut it at the point level  -- Florian Kroessbacher gmail: florian.kroessbac...@gmail.com On Fri, Feb 14, 2014 at 1:30 AM, Jason Aarons (AM)