Hello,   I'm having some issues with the "unlocated phones" report from CER v11.5.4.50000-6.  I have many phones in the report with the status of "Switchport Configuration" under the " Why this ERL is used?" column.  Problem is that all the phones listed with this status are in the ERL db and all information for these phones are correct and valid between the ERL db and the CUCM information.  The switches which the phones are attached are also properly configured.  In fact some of the phones on a single switch will show in the unlocated phones report while other phones on the same switch do not.  Yes, even after running a switch update many times...  All phones are same model and switches are also same model, supported, and code rev.  It's starting to look like the "unlocated" phones report is static (14 pages worth) and never gets any smaller ever after making corrections; the report only grows.   I'm not finding any bug reports on this so far.  If anyone knows what might be going on here or has seen this, please let me know.     Also, is there any possible way to flush out the "unlocated phones" report ? ... View more

I have my CUCM 11.5 directory syncing once a week, but the CUCM is sending a bind (auth) request to the LDAP server about every 2 seconds.  The DC is logging to a Splunk server and the shear number of bind/unbind request from the CUCM to the LDAP server is causing the Splunk license, measured by throughput, to be continuously violated.  If I restart the CUCM DirSync service, the CUCM stops sending the bind requests until the next scheduled sync cycle and then starts its exponential bind requests all over again; Splunk log entries from this issue can reult in as many as 400,000 over a 10 minute interval.  Has anyone seen or experienced this problem?  If so, what was done to correct? ... View more

Hello,   I'm trying to write sip profile that truncates the 10-digit Diversion number dow to the last 4-digits and re-writes that number back into the Diversion header, but I'm not very succesful.  I've written may copy and writes to different headers, but never had a need to first truncate a part of the header and re-write.  For example (IP addresses changed for privacy):   I need the original Diversion header of           Diversion: <sip:1234563333@192.168.0.1:5060;user=phone> To be changed to:           Diversion: <sip:3333@192.168.0.1:5060;user=phone>   Can someone help with the profile rule to make this happen?   Best,     ... View more

I’m having a trouble with modifying my headers and getting it to work on my CUBE outbound sip profile. According to my “debug ccsip message” out put, my header modification is correct, but the inew INVITE isn’t matching my outbound dial-peers. I’m trying to copy the “Diversion” header, from an inbound call, to the “To” header; this works, the new INVITE shows the correct, copied Diversion of \<sip:1234567891@addressofCUCM;user=phone>\". I have dial-peers facing the CUCMs with destination patterns of which the new INVITE should match (ex: destination-pattern 123…….), the CUCM side is setup correctly to receive and process the digits, but they’re not being delivered by the CUBE gateway. This is driving me crazy. Why can’t I get the modified “To” header to match my CUCM facing dial-peers and deliver the new digits? Here are the details: response ANY sip-header Allow-Header modify "UPDATE," ""  request INVITE sip-header Diversion copy "<sip:(.*)@(.*)>" u01  request INVITE sip-header To modify "<sip:5551215555@(.*)>" "<sip:\u01@\1;user=phone\2>"  request INVITE sdp-header Audio-Attribute add "a=ptime:30"  request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"  response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30" INPUT header to modify (invite from service provider): INVITE sip:5551215555@10.253.2.1:5060 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKudo4390058ruj4ghlso0.1 From: "Unavailable" <sip:3233969119@x.x.x.x;user=phone>;tag=2005487715-1532903387839- To: "User 5551215555"<sip:5551215555@10.253.2.1;user=phone> Call-ID: BW222947839290718879647955@Q1 CSeq: 873773408 INVITE Contact: <sip:x.x.x.x:5060;transport=udp> P-Asserted-Identity: "Unavailable" <sip:3233969119@x.x.x.x;user=phone> Privacy: none Diversion: <sip:1234567891@x.x.x.x:5060;user=phone>;privacy=off;reason=no-answer;counter=1 Supported: timer Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,NOTIFY,PRACK,OPTIONS OUTPUT (modified header): INVITE sip:5551215555@10.2.70.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.2.70.9:5060;branch=z9hG4bK9152D4 From: "Unavailable" <sip:3233969119@10.2.70.9>;tag=44B18B87-1977 To: <sip:1234567891@10.2.70.12;user=phone> Date: Sun, 29 Jul 2018 15:29:47 GMT Call-ID: BC3A64C2-92B511E8-B346AE18-324A0900@10.2.70.9 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE:  1800 Cisco-Guid: 3157893444-2461340136-3007360536-0843712768 User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1532903387 Contact: <sip:3233969119@10.2.70.9:5060> Call-Info: <sip:10.2.70.9:5060>;method="NOTIFY;Event=telephone-event;Duration=2000" Expires: 1800 Allow-Events: kpml, telephone-event Max-Forwards: 67 P-Asserted-Identity: "Unavailable" <sip:3233969119@10.2.70.9> Diversion: <sip:1234567891@10.2.70.9>;privacy=off;reason=no-answer;counter=1 Session-Expires:  1800;refresher=uac Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 285 NOW TRYING: Jul 29 22:29:47.953: //189909/BC39A144B340/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.70.9:5060;branch=z9hG4bK9152D4 From: "Unavailable" <sip:3233969119@10.2.70.9>;tag=44B18B87-1977 To: <sip:1234567891@10.2.70.12;user=phone> Date: Sun, 29 Jul 2018 22:29:47 GMT Call-ID: BC3A64C2-92B511E8-B346AE18-324A0900@10.2.70.9 CSeq: 101 INVITE Allow-Events: presence Content-Length: 0 Jul 29 22:29:47.956: //189909/BC39A144B340/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.2.70.9:5060;branch=z9hG4bK9152D4 From: "Unavailable" <sip:3233969119@10.2.70.9>;tag=44B18B87-1977 To: <sip:1234567891@10.2.70.12;user=phone>;tag=436422~edc4535b-0860-4aaa-b294-a1dce0d77157-40417867 Date: Sun, 29 Jul 2018 22:29:47 GMT Call-ID: BC3A64C2-92B511E8-B346AE18-324A0900@10.2.70.9 CSeq: 101 INVITE Allow-Events: presence Reason: Q.850;cause=1 Server: Cisco-CUCM11.5 Session-ID: 00000000000000000000000000000000;remote=9096a4ce9c545503247c7b7dab436422 Content-Length: 0 DIAL PEERS - OUT TO CUCM: dial-peer voice 4001 voip  description Incoming Service provider - CUBE side to UCM  preference 1  destination-pattern 123.......  session protocol sipv2  session target ipv4:10.2.70.12  voice-class codec 2    voice-class sip asymmetric payload full  voice-class sip asserted-id pai  voice-class sip privacy-policy passthru  no voice-class sip early-offer forced  voice-class sip profiles 1  voice-class sip bind control source-interface GigabitEthernet0/0/0  voice-class sip bind media source-interface GigabitEthernet0/0/0  dtmf-relay rtp-nte sip-kpml sip-info sip-notify  ip qos dscp cs5 media  ip qos dscp cs4 signaling  no vad So, what I'm I missing?  Thanks for any help/comments ... View more

Not sure if this is a bug or not, but whenever I try to enable Flexconnect Local Auth at the SSID level, I get the following "The following errors occurred while updating the WLAN: Flexconnect Local Authentication is not supported with Local Switch Central DHCP."  I do have local switching on the Flexconnect, but I do not have "Switch Central DHCP."  I then created a Flexconnect Group and enabled "Enable AP Local Authentication"and No Central DHCP, then palced the APs into that Flexconnect Group, but still seeing "%DOT1X-3-INVALID_REPLAY_CTR" in the WLC syslog while clients roam. I have two questions: Why can't I enable Flexconnect Local Auth per SSID? How can I determine that Flexconnect Local Auth is actually enabled and working?   WLC version is 8.3.122.0 (multiple controllers 8510s and AIR-CT2504-K9S - happens on all them) 3 SSIDs, two in Flexconnect mode APs receiving DHCP from local switch Flexconnect clients receiving DHCP from same local switch Clients for tunnel SSID receiving DHCP from central server   Stumped.  Anyone have a suggestion?   Best   Clients ... View more

I'm creating tunnel interfaces to support up to a potential number of 2000 end point terminations.  I have "heard" that there is an NHRP limitation of 254 per interface, but cannot find a specific reference.  Is this a real and hard limit? Does it depend on the Cisco platform, the tunnel interface, the physical interface?  Any reference would be much appreciated. Thanks ... View more