If I do show run, I still see this:
AP Name : AP-03-Leeds
Admin State : Enabled
AP Mode : FlexConnect
AP Submode : None
Location : Leeds
Reboot Reason : Reload command
Primary controller name : main-wlc
Primary controller IP : 172.11.11.1
Secondary controller name :
Secondary controller IP :
Tertiary controller name :
Tertiary controller IP :
AP join priority : 1
IP Prefer-mode : IPv4
CAPWAP UDP-Lite : Unconfigured
Last Joined Controller name: WLC2504
DTLS Encryption State : Disabled
Discovery Timer : 10
Heartbeat Timer : 30
CDP State : Enabled
Watchdog monitoring : Enabled
IOX : Disabled
RRM State : Enabled
LSC State : Disabled
SSH State : Enabled
AP Username : dmitry
Session Timeout : 300
Extlog Host : 0.0.0.0
I am unable to browse the AP via it's IP address and there is no CiscoAirProvisioning SSID visable either.
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I have a Cisco 1815 that was purchased not too long ago and originally it was joined to a WLC 2504 controller. However, now I want to move it to a different site and want to convert it back to Mobility Express controller. I have followed a few guides online but the AP still starts in a controller mode.
The AP is on a different subnet, different location altogether now by the way, so it is unable to see the original controller.
AP Running Image : 8.7.106.0
Primary Boot Image : 8.7.106.0
Backup Boot Image : 8.5.120.0
Primary Boot Image Hash:
Backup Boot Image Hash:
AP Image type : MOBILITY EXPRESS IMAGE
AP Configuration : MOBILITY EXPRESS CAPABLE
1 Gigabit Ethernet interfaces
2 802.11 Radios
Radio FW version : f1e77cf8ab1e497b17ad53633866ea42
NSS FW version : not available
Any suggestions, please?
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Hello,
I've setup a call handler in Unity Connection 10.x, which accepts the external calls and on Monday/Tuesday it plays a message, then hands up. On other days it meant to let the calls through to a hunt group, instead it just hangs up or try to route to a voicemail box.
I have enabled greetings and transfer rules for both closed and normal. For closed, I set it to transfer to an extension number.
What am I doing wrong here?
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It's a Cisco 2921 router.
During the call I can see that it's using G729r8 codec:
Oct 23 14:37:00.404: //242771/24C17A800000/SIP/Call/sipSPIMediaCallInfo: Number of Media Streams: 1 Media Stream : 1 Negotiated Codec : g729r8 Negotiated Codec Bytes : 20 Nego. Codec payload : 18 (tx), 18 (rx) Negotiated Dtmf-relay : 6 Dtmf-relay Payload : 101 (tx), 101 (rx) Source IP Address (Media): 10.10.10.210 Source IP Port (Media): 30412 Destn IP Address (Media): 88.x.x.x Destn IP Port (Media): 20854 Orig Destn IP Address:Port (Media): [ - ]:0
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Using unicast and the CUBE isn't set up an MTP or Transcoder. The codec used is G729.
I will send the call logs shortly, once everyone leaves the offices to make it easier to log my own call.
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Hello,
We have a CUCM 10, which is configured to work with a CUBE and the external calls are made via a SIP trunk. The music on hold works fine for internal calls that are put on hold, however the external calls are only getting a beep when put on hold.
Any ideas?
Dmitry
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It looks like the DECT phone (BT Xenon and also tried BT Graphite) do not work with the gateway but a standard analogue phone works. Reading more online, it points to an off-hook issue as the gateway cannot detect if the phone if off hook.
If I plug the DECT phone in and try to call it from outside, the phone displays a call coming in and if I try and answer it, nothing happens. Then the standard phone works fine.
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Hello,
I have a Cisco SPA232d (updates to latest f/w), which is connected to a SIP provider. I am want to connect a standard DECT phone to the FXS port on the SPA232d and make/receive calls. I've managed to get the SIP part working and that's all registered but I cannot make calls using the phone plugged into the FXS port. If I plug a basic corded phone into the FXS port, then that works fine... what can be the problem?]
Cheers,
Dmitry
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Picked up a new wireless AP today on NFR to play around with. It's a Cisco 1702i AP in a lightweight mode. I am trying to login to the AP to convert it to autonomous mode but to no avail.
I have tried using a console cable and get nothing (no login screen) and with SSH it does not accept Cisco/Cisco. I've tried resetting it to factory defaults by holding down the mode button while plugging in the power but still nothing...
The console cable works fine on other cisco devices like my ASA and 887 router.
Any ideas?
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It's a lower version it seems Cisco IOS XE Software, Version 03.14.01.S - Standard Support Release Cisco IOS Software, CSR1000V Software (X86_64_LINUX_IOSD-UNIVERSALK9-M), Version 15.5(1)S1, RELEASE SOFTWARE (fc1) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2015 by Cisco Systems, Inc. Compiled Sun 01-Mar-15 03:58 by mcpre
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Hello, I am wondering if anyone has been able to activate CUBE services on a CSR 1000V router? I have it installed on a ESXi 5.5 and acitvated a demo 60-day licence for an AX services. In the Cisco documentations, it says I just need to type in "voice services voip" but the router does not recognise the command. License Level: ax License Type: Paid Subscription Next reload license Level: ax cisco CSR1000V (VXE) processor (revision VXE) with 2151294K/6147K bytes of memory. Processor board ID 9099OPNSUC0 3 Gigabit Ethernet interfaces 32768K bytes of non-volatile configuration memory. 3988460K bytes of physical memory. 7774207K bytes of virtual hard disk at bootflash:. Any ideas?
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I have an IPSec Site-to-Site VPN between two ASA5505 firewalls. This is the current network layout: (Site 1) CME -> SW -> ASA -> INTERNET <- (Site 2) ASA <- SW <-CUCM/Unity Site 1: Data: 192.168.13.0/24 Voice: 10.1.1.0/24 Site 2: Data: 192.168.10.0/24 Voice: 10.10.10.0/24 CME on site 1 is the gateway and does all the routing but the ASA faces the internet. CME points to the ASA for all the internet traffic. Site 2 has ASA as the router and that does all the routing. The problem I am facing is that Site 1 voice network cannot access the voice network at Site 2. I've had a Cisco TAC engineer check everything with VPNs on both sites and he says the problem is down to routing at site 1 but could not check the router or the switch as we only have smartnet on the ASA. This is the error I get when I try to ping 10.10.10.101 (CUCM) from voice network at Site 1. May 24 2015 18:11:12: %ASA-6-110003: Routing failed to locate next hop for icmp from NP Identity Ifc:10.1.1.250/0 to voice:10.10.10.101/0 May 24 2015 18:11:22: %ASA-5-111008: User 'enable_15' executed the 'ping voice 10.10.10.101' command. May 24 2015 18:11:22: %ASA-5-111010: User 'enable_15', running 'CLI' from IP 192.168.13.141, executed 'ping voice 10.10.10.101' I have other sites that have VPNs to Site 2 and have voice network access and that all works perfectly fine. Any ideas?
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