Cisco NetPro -

Multiple CMEs

Sat, 7 Nov 2009 08:08:12 PST

Hello everyone, I have three sites each running CME. I want to allow each site to dial to the other. There are firewalls at each site, but I have allowed RTP and other voice-related protocols and signaling between the sites. When I try to call, I get busy tones. I get a disconnect cause=38 which tells nothing much more than a network fault but I'm not sure exactly what's the fault. I tried different codecs but still no luck. Following are the dial-peers applied: ! voice service voip h323 no h225 timeout keepalive call start slow ! SITE-A Configuration: (Extension range: 7XX and 8XX) ! dial-peer voice 6661 voip description <> destination-pattern [12].. session target ipv4:10.10.5.1 progress_ind setup enable 3 codec g711ulaw ! ! dial-peer voice 6662 voip description <> destination-pattern 3.. session target ipv4:10.10.4.1 ! SITE-B Configuration: (Ext. Range: 1XX and 2XX) ! dial-peer voice 6661 voip description <> destination-pattern [12].. incoming called-number [78].. session target ipv4:10.10.5.1 progress_ind setup enable 3 codec g711ulaw ! ! dial-peer voice 6662 voip description <> destination-pattern 3.. session target ipv4:10.10.4.1 ! SITE-C Configuration: (Extension range: 3XX) ! dial-peer voice 6661 voip description <> destination-pattern [12].. session target ipv4:10.10.5.1 progress_ind setup enable 3 codec g711ulaw ! dial-peer voice 6661 voip description <> destination-pattern [12].. incoming called-number [78].. session target ipv4:10.10.5.1 progress_ind setup enable 3 codec g711ulaw ! ! Any ideas? Thanks, Ahmed

CUCM 6.1.4 Replication state 4 and 3

Sat, 7 Nov 2009 04:10:17 PST

Hello all, I am having replication state 4 with one subscriber and 3 with al the others. I have tried utils dbreplication reset and repair, each time after utils dbreplication stop in all subs and then in pub. Any idea in how to recover? One sympton, is that the same list of servers does not appears in the result of show tech dbstateinfo in all the servers. Thanks for your help

\+! route pattern

Sat, 7 Nov 2009 03:29:15 PST

Please light me how to use + route pattern, when ever I am trying + route pattern, the lady says your call cannot be completed as dial, so my client not able to dial the missed International call directly. I have check the + route pattern without partiton also. Please correct me what I am missing in + route pattern.

VG224 clid on h.323

Sat, 7 Nov 2009 03:19:37 PST

HI, 'm running a vg224 directly connected to a service provider h.323 trunk, calls works well, ut i can't send caller id; i tried using clid command and translation rules, but it doesn't work.

CME with Password (FAC)

Sat, 7 Nov 2009 02:41:34 PST

Hi, I need to enable password for International calls in CME. Is there any way i can enable this feature on CME. Regards, Sunish

How to change the incoming calling number type "unknown" to "Subscriber".

Fri, 6 Nov 2009 22:53:24 PST

The Incoming number from PSTN as below, I want to change the incoming number type from "Type:Unknown" to "Type:Subscriber", please light me how to change the incoming number type as we wish. ----------------------- Calling Party Number i = 0x0080, '25358803 Plan:Unknown, Type:Unknown -----------------------

In CME how to change the numbering type of outgoing calls

Fri, 6 Nov 2009 22:52:21 PST

In CME I am having dial-peer as below when I giving the numbering type for the below dial-peer it is giving busy tone, but when I am removing the "numbering-type international" this dial-peer is matching , please light me how to update the numbering type in CME. --------------- dial-peer voice 10 pots destination-pattern 91407.T numbering-type international forward-digits 10 ------------- port 0/3/0:23

how to enable multicast MoH in local router for SRST

Fri, 6 Nov 2009 21:23:52 PST

please see attached it works in normal mode but failed in SRST, anyone can help ? thanks Attachment Keywords : 1) showconf.txt

CUCM, SRST & Unity Connection

Fri, 6 Nov 2009 21:08:13 PST

Hi We have a scenario, where SRST Router and Unity Connection are in one site, which connects to CUCM in other site. In case the WAN link fails, how will the Unity Connection work with the SRST? How will the Unity Connections ports register with SRST? Is there any configuration which can be performed to use Unity Connection during the WAN failure for the IP Phones registered with the SRST router? Thanks.

7906 phone with CME 4.1 -Login soft key not displayed-

Fri, 6 Nov 2009 17:32:59 PST

Hi, I have CME 4.1 c3845-adventerprisek9-mz.124-15.T6.bin. I have already enable the option call blocking, all the files necessary for the phones are loaded in CF including localization files. I have also tried with template for the phones but the "Login" soft key is not shown. the 7906 phones are new. what else could I check in order to get the softkey "Login" displayed? regards, Attachment Keywords : 1) cme4.1.txt

IOS conference configuration for 32 participants. Need advice.

Fri, 6 Nov 2009 15:39:01 PST

I want to use IOS conference. Where I can read about this? What I need to read and configure to use IOS conference for 32 participants? (IOS version,DSPs, etc) Thank you!

Attendant Console Transfer issue

Fri, 6 Nov 2009 14:23:11 PST

Hello, I'm wondering if anybody has the following issue. We have attendant console (legacy AC and not the new Arc Express client) installed and running on several computers. Call Manager version is 7.0.2 and AC console version is 7.1(1_a). The reason we can use Attendant console is because the call managers were upgraded from 6.1. Everything works fine except the following: When trying to do a blind transfer (on the AC client) of any incoming call from the outside world to a CTI-RP DN that is registered to an IPCC server (a jtapi trigger), the call fails and the outside person hears a busy tone. The same transfer works if the call is internal. Also, the transfer works if it's performed on the phone itself (for both internal/external calls). The kicker is that the consult transfer works on the AC client, and that's what's being used currently by the operator, but that's not ideal as the beginning portion of the ACD greeting gets cut off if the transfer is not completed fast enough. Customer tells me that this was working initially when deployed and then it broked down. The engineers in charge of the IPCC tell me nothing has changed on their end. If it helps any, transfers to non-registered CTI-RP DNs work fine. It only has to do with DN's that are registered as a JTAPI trigger on the IPCC server. I wonder if anyboyd has run into this, or if someboyd has a similar setup and can give it a test and let me know what they get. Thanks in advance, Joseph

Customize Greeting while Call Transfer or Call hold

Fri, 6 Nov 2009 14:21:59 PST

hi all, I have customize greetings (AA) for incoming calls on Unity Connection 2.X. But I dont know how to customize greetings while the call is transfer or while the call is on hold ??? can any tell me how to do it ... Regards,

Installing UCM 6.x on HP equivalent server vs MCS server

Fri, 6 Nov 2009 14:01:27 PST

Hello all, I have a couple questions regarding installing UCM 6.x on HP equivalent server vs MCS server. Hope someone can help clearify. - Are there any additional steps or differences as far as the installation with HP equivalent server ? - Has any one installing UCM 6.x or 7.x on HP equivalent server or non-MSC server ? Please share your experience. Thanks in advance !!! I appreciate any inputs/suggestions !!! Danny

Another IP Communicator one way audio anomoly

Fri, 6 Nov 2009 13:52:46 PST

Hello, I just installed IPC 7.0.3 on a system with a non-supported VPN. The VPN does create a virtual NIC and is assigned an IP address from the LAN subnet so it does work (for the most part). My situation is a little different than other posters I've seen. Here are my scenarios: 1. IPC on the LAN = all works fine. 2. Outside calls over VPN (PSTN to IPC or IPC to PSTN) = all works fine. 3. Inside calls over VPN (IPC to Cisco handset) = one way audio. Handset user doesn't hear IPC user but IPC user hears handset user. The interesting thing is the IPC caller can leave a voicemail message if the handset user doesn't pick up. I've installed the IPC Administration Tool and enabled HTTP access. The getIP.asp file is accessible. I also put the URL in the IP Address Auto Detection URL field. Because the audio from the IPC (via VPN) can get to the Unity server but not to the handset user, I'm not sure I understand how this could be a firewall issue. Any ideas? Thanks in advance. Dan

can't find debug isdn in the as5330

Fri, 6 Nov 2009 13:50:15 PST

Hello i have as5330 with c5350-is-mz.123-17a.bin i can't find debug isdn i was search many documentation please help how can i debug isdn Thank you

CUCM 6.1(2) device unregistered reason code

Fri, 6 Nov 2009 13:44:01 PST

Can anyone tell me what reason code 13 means with reagards to a device unregistering? I suspect that it is network related but can only find reason code 12 on cisco.com. From app logs: ccm: 403998: Nov 05 05:01:06.808 UTC : %CCM_CALLMANAGER-CALLMANAGER-3-DeviceUnregistered: Device unregistered. Device name.:SEPxxxxxxxxxxxx Device IP address.:xxx.xxx.xxx.xxx Protocol.:SCCP Device type. [Optional]:436 Device description [Optional].:phone-xxxx Reason Code [Optional].:13 Cluster ID:StandAloneCluster Node ID:xxxxxxxxxx

QRT - Can this feature be really good for helpdesk?

Fri, 6 Nov 2009 13:36:58 PST

QRT - Can this feature be really good for helpdesk? How I can configure it? I trieed to add softkey to IP phone - but this button doesn't do anythinq. What I need? Thanks.

UCM Upgrade Failure

Fri, 6 Nov 2009 13:35:01 PST

I have 1 pub / 2 subs. I am upgrading from 6.1.1.2000-3. I upgraded my pub to 6.1.3.1000-16. There were no problems. Attempting to upgrade a sub results in error. The following events look suspicious and were logged: CCMInstall|(CAPTURE) Error in ONTAPE RESTORE [Backup data version is not compatible with Online version:| CCMInstall|(CAPTURE) ]| CCMInstall|(CAPTURE) Error in ONTAPE RESTORE [IBM Informix Dynamic Server Version 10.00.UC5XA6 | CCMInstall|(CAPTURE) ]| CCMInstall|(CAPTURE) Error in ONTAPE RESTORE [Physical restore failed - function Invalid archive tape failed code -1 errno 0 | CCMInstall|(CAPTURE) ]| Any ideas?

CUCM 6.1.4 Subscriber fully removent

Fri, 6 Nov 2009 13:26:39 PST

Hello all, I need to fully remove a subscriber from the cluster, so that the entry in db replication info is fully removed, and the license is free for a new subscriber. Do you think it is posible? Many thanks

CUCM 7.1(3) TFTP Auth Fail

Fri, 6 Nov 2009 13:26:22 PST

Since upgrading to CUCM 7.1.3.10000-11 any new phone plugged into the system gets an "auth fail" from the TFTP server when attempting to upgrade the firmware to the version that comes with 7.1(3). Phones that were registered prior to the upgrade work just fine. I am seeing the same behanviour on our non-production environment which is also at 7.1.3-10000-11. Phone Firmware for 7.1(3) on 7961G set is SCCP41.8-5-2SR1S Has anyone sle seen this behaviour before...is anyone else running 7.1(3) yet?

CUCM 6.1.4 User details page blank

Fri, 6 Nov 2009 13:11:39 PST

Hello all, Don´t know why, suddenly when I click in a user, in end users page, nothing appears, but if I create a new one and try to edit it, it is shown properly. II have restored a previous backup and restarted the servers, and still the same problem. This worked before and do not know why happens suddenly. Have anybody seen this before? Thanks

Lost RingTones

Fri, 6 Nov 2009 13:05:03 PST

We recently upgraded to CCM 6.1.4 from 6.1.2 and I have some users who have lost their ring tones. When they try to select them they get "error bad ring tone file"

Unity Connection 7.0 - Digital networking with a unity 4.0(5)

Fri, 6 Nov 2009 11:30:15 PST

Hi, I'm wondering if this is possbile ? I've read a couple of docs and it seems that this option is not possbible unless you set up VPIM. regards

Can I send text message from Cisco IP phone to other phone?

Fri, 6 Nov 2009 11:13:46 PST

Can I send text message from Cisco IP phone to other phone?

Problem ot install CME

Fri, 6 Nov 2009 11:07:26 PST

I use this link http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeinstl.html and I have a problem on step 6:Builds XML configuration files required for SCCP phones. R4(config-telephony)#create cnf-files CNF file creation is already On Updating CNF files %Error deleting flash:SEPDEFAULT.cnf (No such file or directory) %Error deleting flash:XMLDefault.cnf.xml (No such file or directory)CNF-FILES: Clock is not set or synchronized, retaining old versionStamps CNF files update complete Can somebody advice me how to resolve this problem? Thank you!

Phone proxy

Fri, 6 Nov 2009 10:43:15 PST

What's ASA phone proxy solution for remote phone across the internet?

Cisco Call Manage as a DHCP Server- IP Phones are not registered.

Fri, 6 Nov 2009 10:27:03 PST

HI, I have two site , site 1 and site 2. both the sites are interconnected via MPLS .both the sites are using the same IP Address range... 192.168.1.x and I have only one Call Manager for both the sites (which is a DHCP Server for both sites). IP Phones are facing conflicting problem via IP assingment. Please find the attached file for reference. Please suggest the best way out to solve this issue. Thanks, Manish. Attachment Keywords : 1) CallManager as DHCP Server..pdf

CUE7 - stops at boot-loader when reload

Fri, 6 Nov 2009 09:29:54 PST

Hi, when reload the CUE module always stop at boot-loader but able to loading via "boot disk" command and works fine. why CUE module always stop at boot-loader and any way to continut the booting? Is there similar concept to Router's "Configuration register is 0x2102"? Advise please,

PSTN one way calling issue

Fri, 6 Nov 2009 09:27:37 PST

Hello, What could be the concern for one way calling issue between IP communicator & PSTN scenario: cust when using the IPC thru his WAN connection( MPLS cloud) or Wifi connection when in office located in Russia to UK;cust has one way calling issue When he connects VPN secure Client R56 NG, which has secured tunnel till UK where callmanager,GW r placed then customer still complains for one way calling issue for PSTN Line Now when cust access thru UK location, then he donot have one way calling issue with PSTN Note: When customer tried in Russia office, then his other colleague aslo tried,his collegaue hsd similar issue. Could upgrading the VPN client to latest version could be helpful? or do i need to check the configuration on GW side, as since i donot have access to GW so i need proper justification to get it, pl suggest Navin