[OSL | CCIE_Voice] Failed Voice Lab

2013-02-01 Thread Ben John
Guys, i need help in these area: CUCM and CUCME Voice Gateway and Signaling Dial-Plan and Call Routing Features Codec Selection and CAC Voice-Mail Integration High Availability Features During all my four attempts everything works right for me. i was able to make a call even my SRST works fine b

[OSL | CCIE_Voice] [OSL | Automated Alternative Routing]

2013-02-01 Thread ie ravindra
Hi Mates, As per my understanding AAR is triggering when if CAC enabled and if for some reason that call cannot be completed. Therefore we need route that call to the PSTN. The mandatory requirement is the both extensions must register to the same call manager clusted. If Caller A calling to User

Re: [OSL | CCIE_Voice] [OSL | Automated Alternative Routing]

2013-02-01 Thread Justin Carney
Yes, AAR is triggered on CAC reporting out of bandwidth. (Side note - the phone will display "Network Congestion. Rerouting" and this is a service parameter that can be customized, in case that is part of the question requirement.) You are also correct that both phones must be registered to the s

Re: [OSL | CCIE_Voice] [OSL | Automated Alternative Routing]

2013-02-01 Thread Justin Carney
I noticed a typo in that last email (sorry, I clicked "send" too soon) - in the 3 options for H323 digit manipulation, I said num-exp will be between the inbound voip dial peer and outbound *VOIP *dial peer...*the outbound dial peer is POTS* in this case, not voip. (using two voip dial-peers on bo