I'm facing a strange issue on a SIP gateway. We received traces from our provider about calls loops. It appears, if we call the main number, this call is directly send back by the gateway to the sip provider and the loops begin.
Call flow of a working calls:
Sip provider --> Cube/Gateway --> CUCM --> IP Phone
Based on my observation, it's like the call never user the configured translation rules.The number is presented with number 0161413 and have to be transform to +43161413
I add some translation rules but it's never used and calls go out to the SIP provider via dial-peer 150 (see attached config).
I add a rules and dial-peer for test but it's the same see below the modifiy config part
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The short answer is that you don't.... That isn't entirely true while at
the same time it kind of is, but for the most part you don't configure
the softkeys. You enable or disable them via TCL. Here is the long
answer. Be sure to read the whole thing or e...
Topology: IP Phone > Switches > Microsoft NPS setup to forward 802.1x
proxy to > ISE 2.1 patch 3 Authentication: EAP-TLS using Cisco MIC SANs
Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
Cisco 3905 Y Y N Cisco 6911 Y Y N Cisco ...