I've recently embarked on my first journey into Cisco telephony. I have made lots of progress, and most things work fine apart from I so far have not worked out how to forward external calls from my SIP provider to CUE after the timeout. Ringing internally from extension to extension works fine and I can leave voicemail with no problems. When I call in from externally over the SIP trunk, what seems to happen is, after it reaches the No Answer timeout, there is a slight pause, then it starts ringing again, for what seems to be as long as the timeout for a second time, and then I get what I think is referred to as the Fast Busy tone.
I've seen in docs and other posts that I need to have "no supplementary-service sip moved-temporarily" enabled which I have done, and I've tried lots of things but still no joy with this.
Throughout all my research though, I still have slight gaps in my knowledge about how things all fit together so I wanted to ask a few questions that are bugging me and maybe armed with that info I'll work it out anyway, so here goes.
I don't totally understand how an external number ties into the config. I have seen dual-line ephone-dn configurations, where it's possible to specify the internal extension and also public number to an ephone-dn, and I've also seen translation rules to translate the incoming number to an extension. I don't really get whether I should do one or the other, or both of thses things. Also, in the incoming dial-peer, I can reference "incoming called-number", so should I do this instead of, or as well as setting a second number in dual-line mode?
Also, when forwarding to voicemail, I've read that I should tranlate the number at that point to the extension of voicemail, and that the original called number is then placed in a different field so that CUE knows which mailbox to use, but this doesn't seem to happen when I look at my debugs.
Would anyone be able to explain just in simple steps how the whole process should work, or know of any docs that just give an overview? I'm looking for something that just says "call comes in, matches X in the config, number re-written to Y" etc. etc. and that would probably be enough for me to work it out for myself.
Hope I didn't ramble too much and TIA for any responses.
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The short answer is that you don't.... That isn't entirely true while at
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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 ...