Cisco NetPro -

7936 & 7937 Conference Stations

Tue, 17 Nov 2009 09:16:56 PST

Does anyone know if it is possible to disable the beeping sound, you hear when someone calls through on your conference phone when you are already on a line? We have a number of users who have started to complain when they are on a conference call, they hear a loud beep like someone is call that conference phone, is it possible to disable this feature?

FXS busy problem.

Tue, 17 Nov 2009 09:11:19 PST

I have a scenario with a fax printer (Konica Minolta c353) connected to one interface VIC3-2FXS/DID (just one of the fxs ports), the calls are forward to VWIC2-1MFT-T1/E1 interface with MGCPAPP (registered in CUCM7.02). Most of the times we need to reboot the router because the fax line stays busy for long periods of time, and this workaround does not work every time. What can be the problema here??? Router INFO: PID: CISCO2811 PID: VWIC2-1MFT-T1/E1 PID: VIC3-2FXS/DID PID: PVDM2-32 PID: PVDM2-16 IOS: (C2800NM-IPVOICEK9-M), Version 12.4(15)XZ2 ROUTER RUNNING CONFIG: controller E1 0/0/0 pri-group timeslots 1-31 service mgcp ! ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! voice-port 0/0/0:15 cptone PT ! voice-port 0/1/0 cptone PT ! voice-port 0/1/1 cptone PT ! ccm-manager redundant-host 10.88.11.1 10.88.21.1 ccm-manager mgcp ccm-manager music-on-hold ccm-manager config server 10.88.31.1 10.88.11.1 10.88.21.1 ccm-manager config ! mgcp mgcp call-agent 10.88.31.1 2427 service-type mgcp version 0.1 mgcp rtp unreachable timeout 1000 action notify mgcp modem passthrough voip mode nse mgcp package-capability rtp-package mgcp package-capability sst-package mgcp package-capability pre-package no mgcp package-capability res-package no mgcp package-capability fxr-package no mgcp timer receive-rtcp mgcp sdp simple Best Regards, Nuno mgcp rtp payload-type g726r16 static ! mgcp profile default ! ! ! dial-peer voice 999010 pots service mgcpapp port 0/1/0 !

E1

Tue, 17 Nov 2009 08:22:27 PST

When i type: show controller e1 on the router, i find the following: E1 0/1/0 is up. Applique type is Channelized E1 - balanced No alarms detected. alarm-trigger is not set Version info Firmware: 20070321, FPGA: 13, spm_count = 0 Framing is CRC4, Line Code is HDB3, Clock Source is Line. CRC Threshold is 320. Reported from firmware is 320. Data in current interval (588 seconds elapsed): 0 Line Code Violations, 584992 Path Code Violations 0 Slip Secs, 2 Fr Loss Secs, 0 Line Err Secs, 9 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs, 2 Severely Err Secs, 2 Unavail Secs Total Data (last 24 hours) 2 Line Code Violations, 85962281 Path Code Violations, 73 Slip Secs, 555 Fr Loss Secs, 1 Line Err Secs, 1418 Degraded Mins, 72 Errored Secs, 0 Bursty Err Secs, 555 Severely Err Secs, 773 Unavail Secs If, 85962281 Path Code Violations make a problem which make the E1 flap and if it can be solved from the router side configuration or must be solved from the telco side

XFER directly to VM on UC7 not working

Tue, 17 Nov 2009 08:16:20 PST

I have CUCM 6.1 configured with my VM profile XXXX and my CTI RP with my XFER to VM Profile with a DN of *XXXX. This worked at one time, but now calling *3917, you get the general UC greeting. RTMT monitor indicates a "Called Number" and "Redirect Number" of 3917, but a "Last Redirect Number" of *3917. If I add an alternate extension to the user of *3917, it works. It seems that CUCM is stripping the * and just sending the 3917. Does the VM profile mask suppossed to change the "last redirected number" or is there a way to have UC route to VM box based on the "Called number" rather than the "Last Redirected" number? THANKS!!!

voice connect and mgcp gateways

Tue, 17 Nov 2009 07:51:27 PST

Hi all, trying to get voice connect to work, unsuccessfully so far.. obviouisly there is a relationship between voice connect and voice access/enterprise feature access. I've found documentation that says VA /EF only works through h323 gateways, is this the case with VC as well. We only have mgcp gateways so.... thanks

RTMT Error

Tue, 17 Nov 2009 07:38:39 PST

After an upgrade from CM4 to UCM5 when we run RTMT we receive the following error when opening the default profile: "Unable to restore configuration! Host name in the configuration is not defined in DB" I am able to create new profiles and they work fine, but the default not working it's a nuisance... any ideas?

Call Recording Verint issue only for internal calls

Tue, 17 Nov 2009 07:19:39 PST

We have an issue with the install of Verint call recorder. We have CUCM 7 and setup DMS on te required phones, created CTI route point (ext4355) and created JTAPI user as described by Verint guide. The recording works from the recorded phone to external calls and from a non recorded phone to recorded phone without a problem. The issue is from a recorded phone to internal non recorded phone. Verint is saying they don't see CTI on those calls. Any ideas?

RTMT "Unable to restore configuration. Host name is not define on DB"

Tue, 17 Nov 2009 07:13:24 PST

Tonight we moved our Publisher and Subscriber to a new location and also we had to changed their ip's. We are experiencing a problem with RTMT. When i log in to RTMT i get the error "Unable to restore configuration. Host name in the configuration is not on DB" Please help, Thanks. Zeek

AS5350XM showing error during boot

Tue, 17 Nov 2009 07:06:50 PST

Hi, my AS5350XM voice gateway is showing error at the very starting of boot process. The error is like it: System Bootstrap, Version 12.2(1r)1, RELEASE SOFTWARE (fc1) TAC Support: http://www.cisco.com/cgi-bin/ibld/view.pl?i=support Copyright (c) 2001 by cisco Systems, Inc. *** Main Memory Write Bus Error *** Access address = 0x0 PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03 Interrupt Ctrl Reg = 0x1082 *** Main Memory Write Bus Error *** Access address = 0x0 PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03 Interrupt Ctrl Reg = 0x1082 *** Main Memory Write Bus Error *** Access address = 0x0 PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03 Interrupt Ctrl Reg = 0x1082 *** Main Memory Write Bus Error *** Access address = 0x0 PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03 Interrupt Ctrl Reg = 0x1082 *** Main Memory Write Bus Error *** Access address = 0x0 PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03 Interrupt Ctrl Reg = 0x1082 Can anyone has any idea on it? Any help would be highly appreciated.

CUCM 4.2 - Number Range reporting

Tue, 17 Nov 2009 06:28:27 PST

I've got a PSTN number range of a 1000 numbers with the dial plan listing a good 800+ dd's being used by either hunt groups or by users. I'm pretty sure some of these numbers are no longer being used but are still configured, is there any reporting tools which can tell me dd numbers which haven't routed calls over a time threshold... e.g. last 3 months... Thanks,

Wireless AP design for Wireless IPPhone 7921

Tue, 17 Nov 2009 06:25:42 PST

FOr a client they are asking 40 Wireless IP Phone, with 3 floors, IPT side I am clear in design, but for access point design I am bit confused with standalone and WLC design. Could please light me on Wireless design for standalone AP and WLC controller with AP , in the design plan to give both as options. And also how to confirm a AP is a standalone or it wil work only with WLC. Please light me with wireless design for IPT.

Attendant Console / Pilot Point

Tue, 17 Nov 2009 06:21:29 PST

Hi, Currently I'm trying to install an attendant console and am having an issue when associating the Pilot Point to the Application user in Controlled devices. When clicking on Find More Pilot Points, the pilot point in question is listed however when selecting 'add selected' it simply sits there and does nothing. I can add a physical extension to the controlled device field but not a pilot point! On further investigation I've found the Pilot point is not registered and doesn't have an IP address, could this be causing the issue? If it is how do I get the pilot point registered?! Any pointers would be greatly appreciated!

Cause i = 0x80BF - Service/option not available, unspecified

Tue, 17 Nov 2009 05:34:23 PST

Hi, I have a CME which is connected to a E1 PRI line.Iam able to call outgoing calls from cisco phones but the problem here is incoming calls are not coming. when i did debugging , i got the following error .. Cause i = 0x80BF - Service/option not available, unspecified Can anyone help me out here pls.. regards

CME Billing

Tue, 17 Nov 2009 05:24:28 PST

Hi Guys I m looking for call billing platform for Cisco CME.Call manager Express.I have tried Stonevoice but it has a major limitation in that it cannot capture any calls TRANSFERRED Or FWDed. Any other solution

light me the best voice WAN solutions via SHDSL

Tue, 17 Nov 2009 05:23:43 PST

could please someone compare the 3 SHDSL card: Cisco 1-Port G.SHDSL WAN Interface Card (part number WIC-1SHDSL-V3) Cisco 2-Pair G.SHDSL HWIC (HWIC-2SHDSL) Cisco 4-Pair G.SHDSL HWIC (HWIC-4SHDSL) I need to provide WAN voice a solution for two branch office, between Dubai and Oman, from the above one which SHDSL card can I quote; bit confused with SHDSL connection. Could please light me the best voice WAN solutions between branch office via WAN; 100 users in Dubai and 24 user in Oman.

About Gateway protocols: SIP/H.323/MGCP

Tue, 17 Nov 2009 05:17:42 PST

Which one is widely deployed in real world including enterprise or service provider environments? Is SIP is fully supported on Cisco devices or its still under development ? Thanks, Dev

Second line and Softkey template

Tue, 17 Nov 2009 04:28:36 PST

Hi I have two question and I hope someone can help me 1. When I call DN and the prime line is in use, the ring out tone (of my prime line) to be different for the second line (called DN). Now I have same ring out tone for all calls I made. How to accomplish this? 2. When the phone is in the Ring In State a can’t add End Call in the Softkey Template. Is there any way to bypass this? We have CUCM 6.1.3

How to set up Call Queuing

Tue, 17 Nov 2009 04:09:54 PST

Hi All, I have a couple of users in a line group that I would like to set them up some basic queuing. Is this possible? I have cisco Business edition 6.1.3.2000-1 with cisco unity connection Version 2.1.3ES23.1000-23. thanks in advance, Mike

can CME talk to gatekeeper with G729r8 or it is only g711

Tue, 17 Nov 2009 04:06:36 PST

Hi, I have this network diagram CUCM7---GK---CME7 CUCM & CME are in the same zone to the GK I need to limit the max bandwidth inside the zone to 64k Bu tI found a strange behavior: when calling from CMM to CME,the phones show that it is a g729 call and the o/p of 'sh gatekeeper call' shows that it consume 16K which is expected But when calling from CME to CCM the phones show that it is g729 call but the o/p of 'sh gatekeeper call' shows that the call consume 128K,so the gatekeeper need 128 K for this call although I have configured it to use g729 I have configured the outgoing dial-peer to the GK on the CME with G729 and the call allready match with it,also the DP of the ICT gk controlled is using only g729 Also transcoder is configured on the CME in case the CME will need it So why the CME insist in using 128 K for this call Any feed back will be appreciated thanks

Passwords for Meet-Me Conference

Tue, 17 Nov 2009 02:29:47 PST

hi, My customer wants me to configure password on meet-me conference !! how can we achieve this ?? Regards,

called party number config

Tue, 17 Nov 2009 02:26:00 PST

For my CCIE voice lab pratice, I am using CUCM 7.1 Business edtion, when I try to use called party transformation mask, it fails always, I have update the CSS for called party in device pool and gateway, could please light me what I am missing for called party transformation mask.

PLAR with delay of 6 seconds on FXS ports

Tue, 17 Nov 2009 02:19:31 PST

Hi there Is this possible??? If I lift the analog phones connected to the FXS port on the VG224, I should be able to dail out to any number however if I jlift the phone and keep it off-hook for 6 seconds without dialing any number, then it should dial out to a particular number. is it possible??

ATA problem after UCM Upgrade 6.1 to 7.0.2

Tue, 17 Nov 2009 01:29:40 PST

Hey, we've recently upgraded our UCM cluster from 6.1 to 7.0.2 on new hardware. We changed the tftp-server in the dhcp scopes and made a reset of the phones on the old cluster. after contacting the new tftp-server (new publisher) the phones loaded the new firmware and registered on the new cluster. so far so good, but we've problems with our ata's. They got the new tftp-setting by dhcp but they didn't load the new firmware. It seems, that are no .cnf-files on the tftp for the ata's, because they have a "SEPDEFAULT.cnf" as tftp-file. futhermore the ata's are still registered on the old cluster. As workaround, we've added the new publisher as a static ip directly on the ata (webinterface). So what can i do?

Routing configuration (integration of Cisco CM4.2, Nortel 81c and AVAYA)

Tue, 17 Nov 2009 00:44:21 PST

3 sites: Site A: Cisco CM 4.2 (212-555-75XX) Site B: Nortel 81C (212-555-79XX) Site C: AVAYA (305-666-45XX) History: Site A and B are in the same building, connected via PRI link, routing plan had been configured, user can dial to each other directly by 4 digits extension (75XX and 79XX) Project: Site A and C have IP connection already, recently, we use voice gateway to connect Site A and C, and configured all site A, B and C. When Site A user dials 9-1-305-666-45XX, CM will route the call via voice gateway to site C. (Done) When Site B user dials 9-1-305-666-45XX, Nortel will route the call to CM, and CM will dump all digits to site C via gateway. (Done) When site C user dials 9-1-212-555-75XX, AVAYA will route the call to site A and dump all 10 digits to Cisco CM. (Done) When site C user dials 9-1-212-555-79XX, AVAYA will dump all digits to Cisco CM and CM will route it to Nortel. (Failed, fast busy tone on site C) Troubleshooting: 79XX configured as route pattern in CM. Gateway transmitted all 10 digits number from site C; it may work for site A Cisco CM, but can’t recognize by Nortel. Question: How to configure the Cisco CM or Gateway to only transmit last 4 digits (79XX) to Nortel when received 212-555-79XX from site C? I appreciate your help!

CDR Backup

Mon, 16 Nov 2009 22:04:14 PST

Hi, I am planning a upgrade from 4.x to 6.x call manager. CDR records older than 180 days will be deletedduring DMA import. I dont want to delete CDR records, instead I would like to backup CDR data. How can this be acheived. -prakash

Backup with cdr failure

Mon, 16 Nov 2009 22:02:56 PST

Hi i'm running ccm 4.2 with bars 4.0.14, when i made the backup includding the cdr option it returns an error and only backs up the ccm configuration portion. I suspect this is because of the size of the cdr database, it have not been purged long time ago. Heres is a log sample with and without cdr. Tahnks. Attachment Keywords : 1) Backup without cdr.txt - Backup with cdr.txt 2) Backup with cdr.txt

Cisco phone designer Widgits

Mon, 16 Nov 2009 20:14:48 PST

Anybody tried the phone designer ? i have installed it on a PC and connected to a server running callmanager 6.1.2, managed to set the background screen and change the ringtones no problem. I now come back to the tool after 5 days and try to login, it say internal database error, if i choose an incorrect password it says access denied, so i knoe its talking with callmanager, anybody understand how this thing integrates with callmanager and where i can start to toubleshoot?

AVST CallXpress - CUCM Callmanager Integration: Sample Config Needed

Mon, 16 Nov 2009 17:57:39 PST

Does anyone have an AVST CallXpress voicemail system integrated with Callmanager using the Radvision SCCP (7940) integration? If so, could you post your Callmanager configuration to show how you are getting DID calls to forward to a mailbox? I have the integration up and running, but call forward to personal greeting is not working for me. Forwarded calls answer with the initial call processor mailbox defined in the answer mode config. FYI, we have CallXpress version 6.5 and Callmanger 4.3. Thanks!

Meeting Place conference

Mon, 16 Nov 2009 16:29:40 PST

Hi All, Need help. Calls to Meeting Place conference that present over PSTN lines are unable to attend conferences. Callers report the system is doubling the digits sent causing Meeting Place to deny access to the meeting. Please help me on this issue.

ATA 186 and faxing

Mon, 16 Nov 2009 15:23:44 PST

I have a fax machine that has had an ATA 186 connected to it for over two years without any problems. All of a sudden you could send faxes but not receive them. We moved the fax machine and ATA 186 to another room and tried out a different ethernet jack. The fax machine will send and receive and works like it should. Is this a problem with the ATA 186 or the ethernet connection ? Thanks