Oliverrietbrock, XML configured, I tried both vestion 4.6.x and 4.7.x but even if my PC is IPv6 only, anyconnect crashes. And with dual stack it will always prefer IPv4. So I'm wondering if it is NOT supported, I just haven't been able to find supporting documentation about it.   Rolando A. Valenzuela. ... View more

What type of problems? You cannot dial? No audio? Error message? Use RTMTto collect the logs of the calls that are failing.   https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200787-How-to-Collect-Traces-for-CUCM-9-x-10-x.html   Rolando A. Valenzuela. ... View more

Check this article, may be useful. At the end of the day you just want to define remote MOH servers. https://www.netcraftsmen.com/live-stream-music-on-hold-and-cisco-uc-on-ucs/ Rolando A. Valenzuela ... View more

But why?! hahahah I'm assuming you still want to keep +0123456789 as one of the SNR numbers, right?   I haven't test it but having that incoming call using another Partition/CSS maybe?   Rolando A. Valenzuela.     ... View more

I would not recommend that for multiple reasons, like user experience, service availability, cost, and others. Call coverage is important yes but what you want to setup goes south as soon you think on a third call. What if the two phones are actually on a call and you have a third, where are you going to send it? No Answer doesn't work on that case.   Adding multiple lines with the same members will only extend the hunting period but at some point it will exhaust all available options.   I don't know the user case but it will be better if you handle the call somehow like sending it to VM or a recording like "sorry try again later" or anything... leaving the call there endlessly sound troublesome for me, what if it is a ghost call at night?   --------------------------------- Answering your question, the best way I can think of, will be having two Hunt Pilots and make them to RING each other for the No-Answer condition.   Call -> Hunt A ---(no answer)---> Hunt B ---(no answer)---> Hunt A ---(no answer)---> Hunt B........   Rolando A. Valenzuela. ... View more

If I'm reading this right DP 10 = Incoming call from ITSP DP 20 = From CUBE to CUCM DP 30 = From CUBE to CUCM   I see a two problems with DP 20, you have a translation-profile there and you are also mixing called-number with destination-pattern. I believe the configuration of DP 30 is correct, so try disabling DP20 either shutting it down or increasing the preference and test again.   Rolando A. Valenzuela ... View more

On the logs I see State of The Call : STATE_DEAD and also SIP/2.0 503 Service Unavailable You remove the IPs from the logs (which is fine for security, but next time just replace for something random like 5.5.5.5) so I'm not sure where is CUBE sending those messages but my guess is that it cannot communicate with the SIP provider IP.   Please check that. Rolando A. Valenzuela. ... View more

:( can someone confirm this? ... View more

Huntstop should work, unless there's another condition selecting which dial-peer is used. Can you reproduce this behavior? Can you include debugs showing it?   "A Cisco Gateway only attempt to route a call via one eligible outbound dial-peer at a time. If a failure condition observed on the first selected dial-peer the gateway then attempts to route the call out the next eligible dial-peer. This continues until the call either succeeds or fails because there are no more eligible dial-peers left to try. A common symptoms of dial-peer hunting and failure is a noticeable delay in ringback while making calls. Debugs are usually needed to verify exactly why the call is failing on the first few dial-peers before finally working across another.  The command huntstop can be employed on a dial-peer if an administrator does not want a gateway to look for another dial-peer when a failure condition is observed." Ref: https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html   Rolando A. Valenzuela ... View more

QoS will work only link congestion. If your link it is not suffering congestion, your policy will not be matched, you can check that with the command "show interfaces" and look for the Queue statistics.   Regards. Rolando A. Valenzuela ... View more

Do you have a diagram to understand better? I'm assuming the call is between a Cisco IP Phone (registered with CUCM) and another endpoint (resisted with Asterisk). No matter what, CUCM is the middle man between the Cisco IP Phone and Asterisk, so it will communicate/forward all the different SIP states the call is in, so I'm not sure if the transparency you are looking is possible but Asterisk should be able to understand and do the call setup correctly on its end.   I do not understand the "when the PBX make a call the operator is listening the ringing". What PBX? Who/what made the call? The called phone should not ring?   Regards. Rolando A. Valenzuela. ... View more

If you have configured and correctly associated the regions, there should be only one list to pick from. Now, remember that if the regions are not defined, it will uses system default for intra and inter region calls. Those settings can be found on: System > Service Parameter > Clusterwide Parameters (System - Location and Region) > Default Audio Codec List between Region   Regards. Rolando A. Valenzuela. ... View more

Yes you are, just by adding "allow-connections from-type to to-type" you are enabling the traverse features on the router. So please double check the routing settings to confirm reachability between both phones.   Ref: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-overview.html   Rolando A. Valenzuela ... View more