Hi Tim,    Sorry, I have been out of the office.  Thanks for the diagram, I understand that flow.  The issue is this, one of our servers in the cluster did not have SSO enabled (it was a new build) and login was being denied.  What I don't understand is why wouldn't the client use another server in the cluster?   Once I updated the metadata on the new CUCM node, everything was fine.  So I was asking in the event that we had an issue with a SUB, SSO could potentially break users signing into Jabber since the client doesn't seem to try another node in the cluster.   Does that make sense?!     ... View more

Hi George,   Thanks for the reply.  Not sure why I cannot get my head around this.  When the user calls from the remote office FXS port, they get a fast busy.  I am sure that I am missing something in the dial-peers, here is what I have:   Remote Office FXS connection dial-peer voice 514933 pots  description *** Fax from ATL Main Office ***  destination-pattern 514933  port 0/2/0 dial-peer voice 1 voip  description *** Outbound Fax Calls via the ATL Office ***  destination-pattern 9.T  modem passthrough nse codec g711ulaw redundancy  session target ipv4:10.38.12.1  dtmf-relay h245-alphanumeric  codec g711ulaw  fax rate disable  no vad PRI Gateway Office dial-peer voice 514933 voip  description *** Fax To SCN ATL Office ***  destination-pattern 514933  modem passthrough nse codec g711ulaw redundancy  session target ipv4:10.38.12.1  dtmf-relay h245-alphanumeric  codec g711ulaw  fax rate disable  no vad dial-peer voice 1 pots  description *** Outbound Dial-Peer ***  translation-profile outgoing ATLFaxOut  destination-pattern 9.T  port 0/1/0:23  forward-digits 10 dial-peer voice 2 pots  description *** Inbound Dial-Peer ***  translation-profile incoming xlatextn  incoming called-number .  direct-inward-dial  port 0/1/0:23   I thought that the VOIP dial-peer on the FXS router would send anything the user dials to the PRI gateway, but I know that I missing something on the PRI gateway to handle the leg from the FXS router, hence the busy signal.  Any suggestions?!   Thanks Again,   Corey ... View more

Yes, I am using Nuance.  There is a mixture of menus asking for DTMF key presses, as well as a part of the code that allows the caller to speak (ASR), it seems as though ASR being in the script causes any background noise during a menu/prompt playback to get interrupted, hence breaking the script.   This will happen at any location in the script as long as ASR is involved.  Under the application trigger, the override media termination is set to "yes", and MRCP ASR GROUP #1 is above default (allows the ASR portion of this script to work).   So for example:  The script starts with a welcome message (no barge-in), then the next menu states Press 1 for x Press 2 for y, with barge in set...if I am talking while the prompt is playing asking me the press a button, barge in kicks in even though I haven't pressed any keys. I don't want this portion of the menu to use ASR, so having them say the number will not work.  The ASR comes into play later in the script.   Thanks all! ... View more

Thanks Scott, That is what I thought.  My concern is dialing from HQ if the WAN goes down.  The translation would still be accessible because I am able to hit the CM, the number would be translated to my 6 digits, and fail since the WAN link is down, am I over thinking this? I see how if I am dialing translating to 6 digits from a remote site...with the WAN link down the remote site would not be able to translate and therefore hit a dial-peer to the PSTN.  I cannot seem to understand what happens when a user dials from HQ when the WAN goes down.....sorry to be a pain. Thanka again for you help! ... View more

Hi everyone!  I have a quick question that relates to this post.  I have a my central CUCM site and remote H.323 gateways setup.  The issue now is when users dial the full 10 digits my calls route over the PSTN, not what I want.  So in order to get them to rotue within the WAN I assume the translation pattern takes care of this, as you have noted?  I did some testing and indeed can make a translation pattern work.  So my only question is what happens from my central site if the WAN link goes down?  Will the call then go out the PSTN?  I am afraid if I implement these translations and the WAN link goes down, the call will not connect, am I wrong in assuming this? Thanks all! ... View more