getXML('IP Telephony26001 dvanzee@rgl.netNov 17, 2009, 9:16am PSTDoes 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? nunoscostaNov 17, 2009, 8:56am PSTI 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).<br />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.<br />What can be the problema here???<br /><br /><br />Router INFO:<br />PID: CISCO2811 <br />PID: VWIC2-1MFT-T1/E1 <br />PID: VIC3-2FXS/DID <br />PID: PVDM2-32 <br />PID: PVDM2-16 <br />IOS: (C2800NM-IPVOICEK9-M), Version 12.4(15)XZ2<br /><br />ROUTER RUNNING CONFIG:<br /><br />controller E1 0/0/0<br /> pri-group timeslots 1-31 service mgcp<br />!<br />!<br />interface Serial0/0/0:15<br /> no ip address<br /> encapsulation hdlc<br /> isdn switch-type primary-net5<br /> isdn incoming-voice voice<br /> isdn bind-l3 ccm-manager<br /> no cdp enable<br />!<br />voice-port 0/0/0:15<br /> cptone PT<br />!<br />voice-port 0/1/0<br /> cptone PT<br />!<br />voice-port 0/1/1<br /> cptone PT<br />!<br />ccm-manager redundant-host 10.88.11.1 10.88.21.1<br />ccm-manager mgcp<br />ccm-manager music-on-hold<br />ccm-manager config server 10.88.31.1 10.88.11.1 10.88.21.1<br />ccm-manager config<br />!<br />mgcp<br />mgcp call-agent 10.88.31.1 2427 service-type mgcp version 0.1<br />mgcp rtp unreachable timeout 1000 action notify<br />mgcp modem passthrough voip mode nse<br />mgcp package-capability rtp-package<br />mgcp package-capability sst-package<br />mgcp package-capability pre-package<br />no mgcp package-capability res-package<br />no mgcp package-capability fxr-package<br />no mgcp timer receive-rtcp<br />mgcp sdp simple<br /><br /><br />Best Regards,<br />Nuno<br />mgcp rtp payload-type g726r16 static<br />!<br />mgcp profile default<br />!<br />!<br />!<br />dial-peer voice 999010 pots<br /> service mgcpapp<br /> port 0/1/0<br />!<br /> 30 p.bevilacquaNov 17, 2009, 9:05am PSTTry updating IOS, eg 12.4(22)YB4nunoscostaNov 17, 2009, 9:11am PSTI have another scenario with different fax printer on another location, the router have the same characteristcs.<br /><br />Are you sure the IOS is the problem here?<br /><br />Thanks in advance. EBIC-2007Nov 17, 2009, 3:48am PSTWhen i type: show controller e1 <br />on the router, i find the following:<br />E1 0/1/0 is up.<br /> Applique type is Channelized E1 - balanced<br /> No alarms detected.<br /> alarm-trigger is not set<br /> Version info Firmware: 20070321, FPGA: 13, spm_count = 0<br /> Framing is CRC4, Line Code is HDB3, Clock Source is Line.<br /> CRC Threshold is 320. Reported from firmware is 320.<br /> Data in current interval (588 seconds elapsed):<br /> 0 Line Code Violations, 584992 Path Code Violations<br /> 0 Slip Secs, 2 Fr Loss Secs, 0 Line Err Secs, 9 Degraded Mins<br /> 0 Errored Secs, 0 Bursty Err Secs, 2 Severely Err Secs, 2 Unavail Secs<br /> Total Data (last 24 hours)<br /> 2 Line Code Violations, 85962281 Path Code Violations,<br /> 73 Slip Secs, 555 Fr Loss Secs, 1 Line Err Secs, 1418 Degraded Mins,<br /> 72 Errored Secs, 0 Bursty Err Secs, 555 Severely Err Secs, 773 Unavail Secs<br /><br />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 30 natan.zarhinNov 17, 2009, 4:26am PSTHello there.<br />Check out using "cablelength" command. I suppose framing/line code is correct.. <br />I would contact Telco anyway to check other side of this e1. <br />Natan. EBIC-2007Nov 17, 2009, 4:28am PSTSorry, what is the cablelength command. p.bevilacquaNov 17, 2009, 5:22am PSTYou have been asking this question again, please do not open duplicates.<br /><br />You have a faulty circuit, even if telco says otherwise. EBIC-2007Nov 17, 2009, 5:27am PSTsorry, for annoying you. p.bevilacquaNov 17, 2009, 5:32am PSTIt does not annoys me, it is just good forum etiquette not to open thread duplicates.<br /><br />Remember: forum cannot fix a faulty circuit. Telco can. EBIC-2007Nov 17, 2009, 5:34am PSTok, i understand what do you mean, but, i am this time asking another different question. p.bevilacquaNov 17, 2009, 5:44am PSTIt is a side effect of your E1 flapping.<br />It must be solved by telco. burleymanNov 17, 2009, 7:28am PSTPaolo is correct, this needs to resolved by the telco. I was also having the same issue on one of my E1 circuits and the telco kept saying it was our equipment. Please check the following document for more information and what you can do to show the telco it is on there end.<br /><br /><A HREF="javascript:newWin('http://www.cisco.com/en/US/tech/tk713/tk628/technologies_tech_note09186a00800a70f6.shtml')">http://www.cisco.com/en/US/tech/tk713/tk628/technologies_tech_note09186a00800a70f6.shtml</A><br /><br /><br /><A HREF="javascript:newWin('http://www.cisco.com/en/US/tech/tk713/tk628/technologies_tech_note09186a00800f2fa1.shtml')">http://www.cisco.com/en/US/tech/tk713/tk628/technologies_tech_note09186a00800f2fa1.shtml</A><br /><br /><br /><A HREF="javascript:newWin('http://www.cisco.com/en/US/docs/internetworking/troubleshooting/guide/tr1915.html')">http://www.cisco.com/en/US/docs/internetworking/troubleshooting/guide/tr1915.html</A><br /><br /><A HREF="javascript:newWin('http://www.cisco.com/en/US/tech/tk713/tk628/technologies_tech_note09186a00800f99bb.shtml')">http://www.cisco.com/en/US/tech/tk713/tk628/technologies_tech_note09186a00800f99bb.shtml</A><br /><br /><br />Hope this helps.<br /><br />Mike<br /><br /> EBIC-2007Nov 17, 2009, 8:10am PSTAppreciatted for these great docs. burleymanNov 17, 2009, 8:22am PSTNo problem. Good luck with the telco. Our circuit finally got fixed. I was getting errors every few minutes and getting drops a few times a day while they kept telling me it was my equipment. Then all of a sudden all my errors went away and no more issues with the circuit dropping. When I asked what they did they said they did not do anything, aren't telco's great... :-)<br /><br />remember to rate helpful posts.<br /><br />Thanks,<br />Mike mmertens@iplogic.comNov 17, 2009, 7:57am PSTI 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?<br /><br />THANKS!!! 30 rajeshrevuruNov 17, 2009, 8:06am PSTCan you check if you have alerting name set on *XXXX DN? mmertens@iplogic.comNov 17, 2009, 8:16am PSTHi,<br /> Alerting name is blank. For "Forwarding Call" information for the DN, I have the default "Caller Name" and "Dialed Number". I've also tried unchecking the "dialed Number" and checking "Redirected Number" but I see the same info on UC Port Monitor.<br /><br />Thanks. reidgNov 17, 2009, 3:32am PSTHi 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....<br /><br />thanks 30 iptuser55Nov 17, 2009, 5:40am PSTIf Voice Conenct is Single Number Reach then it does work with MGCP as I`ve just set it up on my cucm. What error are you having, the CSS which is used to ring out an external DN - only external numbers are permitted.Also the CSS the app. uses is to re-routing CSS and not the CSS in the Remote Destination profile, Destination. the numbers you must enter in the RDP has to include your lcoal PSTN access code reidgNov 17, 2009, 5:59am PSTThanks for reply,<br /><br />its now working, or maybe I wasn't giving the external phone long enough to ring, I just assumed this would be simultaneous but there is a delay of a good 3-4 seconds. Can this be tweaked do you know?<br />Also since I activated this and set up, the access list option has disappeared from the 'Device>DeviceSettings' menu, strange, any idea??? iptuser55Nov 17, 2009, 6:06am PSTRing time- Play around with the Delay before Ringing timer in the Remote Destination page set it to zero - should be simultaneous however it also depends on your Route patterns -Any ! settings?. Create a specific Route pattern for an exact match for testing <br /><br />Access List- Not seen this on my one reidgNov 17, 2009, 6:10am PSTthanks, the specific route pattern for testing is a good idea, I'll try that.<br /><br />Access List, are you saying that you don;t have access list function under CUCMAdministrator>device>devicesetting, or that you havn't saw it idsappear before. I'm on a version 7, and it was there, but has now disappeared... iptuser55Nov 17, 2009, 6:45am PSTWhat is your set up as I`ve got 7 in our lab- did you create an access list and now the option is gone or left it as none reidgNov 17, 2009, 7:51am PSTit was there, created an access list, now the menu option is missing, although this access list is available to apply to Remote destinations. carscaddenbJun 23, 2008, 4:38am PSTAfter an upgrade from CM4 to UCM5 when we run RTMT we receive the following error when opening the default profile:<br /><br />"Unable to restore configuration! Host name in the configuration is not defined in DB"<br /><br />I am able to create new profiles and they work fine, but the default not working it's a nuisance... any ideas? 30 tim.gilesJun 23, 2008, 6:46am PSTI know this is a stupid question but have you downloaded the new plugin and installed? descasharedNov 17, 2009, 7:15am PSTHi Carscaddenb,<br /><br />i have the exact same issue, i have upgraded from cucm 4.1 to 6.1 and have the same error message, have you fixed this issue? how you did it?<br /><br />Thanks, descasharedNov 17, 2009, 7:38am PSTHi Carscaddenb,<br /><br />i have the exact same issue, i have upgraded from cucm 4.1 to 6.1 and have the same error message, have you fixed this issue? how you did it?<br /><br />Thanks,s-jordanNov 17, 2009, 7:19am PSTWe 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.<br />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?<br /><br /><br /> eferraros@ewpartners.comJun 26, 2008, 9:37pm PSTTonight we moved our Publisher and Subscriber to a new location and also we had to changed their ip's.<br />We are experiencing a problem with RTMT.<br />When i log in to RTMT i get the error "Unable to restore configuration. Host name in the configuration is not on DB"<br />Please help,<br />Thanks.<br />Zeek 30 javalencJun 27, 2008, 5:56am PSTwhat version?<br /><br />CSCee76271<br />Cannot restore some profile saved<br /><br />1) From the CCMAdmin web page, select Service then Service Parameters<br />2) Specify the CallManager server and select RIS Data Collector<br />3) Ensure that "Data Collection Enabled" is set to True<br />4) Verify the Primary Collector, must be the hostname of the server (PUB recommended) where you installed RTMT<br /><br />HTH<br /><br />javalenc<br /><br /><i>if this helps, please rate</i> descasharedNov 17, 2009, 7:13am PSTHi Javalenc<br /><br />i'm having the exact same issue, just that under service parameters of RIS data collector i cannot see the data collection enabled parameter. in my case i have upgraded from ccm 4.1 to ccm 6.1.4<br /><br />Thanks,bilasheceNov 17, 2009, 12:38am PSTHi, my AS5350XM voice gateway is showing error at the very starting of boot process. The error is like it:<br /><br />System Bootstrap, Version 12.2(1r)1, RELEASE SOFTWARE (fc1)<br />TAC Support: <A HREF="javascript:newWin('http://www.cisco.com/cgi-bin/ibld/view.pl?i=support')">http://www.cisco.com/cgi-bin/ibld/view.pl?i=support</A><br />Copyright (c) 2001 by cisco Systems, Inc.<br /><br />*** Main Memory Write Bus Error ***<br />Access address = 0x0<br />PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03<br />Interrupt Ctrl Reg = 0x1082<br /><br />*** Main Memory Write Bus Error ***<br />Access address = 0x0<br />PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03<br />Interrupt Ctrl Reg = 0x1082<br /><br />*** Main Memory Write Bus Error ***<br />Access address = 0x0<br />PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03<br />Interrupt Ctrl Reg = 0x1082<br /><br />*** Main Memory Write Bus Error ***<br />Access address = 0x0<br />PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03<br />Interrupt Ctrl Reg = 0x1082<br /><br />*** Main Memory Write Bus Error ***<br />Access address = 0x0<br />PC = 0xbfc0a114, Cause Reg = 0x1000800, Status Reg = 0x30410c03<br />Interrupt Ctrl Reg = 0x1082<br /><br />Can anyone has any idea on it? Any help would be highly appreciated. 30 p.bevilacquaNov 17, 2009, 5:46am PSTTry replacing memory module.bilasheceNov 17, 2009, 7:05am PSTIt has three memory cards inside there and tried to boot putting arbitrary two modules, but no luck...is there any probability that all of the three got problem? p.bevilacquaNov 17, 2009, 7:06am PSTNot much probability.oneille1977Nov 17, 2009, 5:57am PSTI'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...<br /><br />Thanks,<br /> 30 javalencNov 17, 2009, 6:28am PSTNo, dates of last usage are not recorded<br /><br />HTH<br /><br />java<br /><br /><i>if this helps, please rate</i><br /><br /><A HREF="javascript:newWin('http://www.cisco.com/go/pdihelpdesk')">www.cisco.com/go/pdihelpdesk</A> bsingaraNov 16, 2009, 8:49pm PSTFOr 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.<br /><br />Could please light me on Wireless design for standalone AP and WLC controller with AP , in the design plan to give both as options.<br /><br />And also how to confirm a AP is a standalone or it wil work only with WLC.<br /><br />Please light me with wireless design for IPT. 30 j.schulenbergNov 17, 2009, 6:25am PSTI would suggest you subcontract this to a partner who is familiar with wireless voice designs. There are a lot of details and caveats to make sure it works.<br /><br />For example: autonomous APs are not viable for voice installations. You need a controller to prevent the roam times from interrupting the call.<br /><br />At a minimum, you should read the Voice over Wireless LAN Design Guide:<br /><A HREF="javascript:newWin('http://www.cisco.com/en/US/solutions/ns340/ns414/ns742/ns820/landing_voice_wireless.html')">http://www.cisco.com/en/US/solutions/ns340/ns414/ns742/ns820/landing_voice_wireless.html</A> juniper76Nov 12, 2009, 2:51am PSTHi,<br /><br />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. <br /><br />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!<br /><br />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?!<br /><br />Any pointers would be greatly appreciated! 30 anis_ciscoNov 14, 2009, 10:58pm PSTHi,<br /><br />First try to registerd the pilot point with your CCM, after configuring the pilot point if you are not able to register it then try by restarting CTI Manager or Attendant Console services !!<br /><br />then you must get the pilot point to associate with application user !!!<br /><br /><br />HTH<br /><br />Regards,<br />Anis juniper76Nov 17, 2009, 6:21am PSTThanks for coming back to me Anis<br /><br />I've now got the pilot point registered by restarting the CTI Manager Service on my publisher. <br /><br />However i still can't associate the Pilot point to the ac user! The pilot point is listed, i tick the box next to it and then add selected but nothing happens! sivakumar.kvpNov 17, 2009, 3:46am PSTHi,<br /><br />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.<br />when i did debugging , i got the following error ..<br />Cause i = 0x80BF - Service/option not available, unspecified<br /><br />Can anyone help me out here pls..<br /><br />regards 30 p.bevilacquaNov 17, 2009, 5:34am PSTPlease always include the full trace.cm@benau.comNov 17, 2009, 4:07am PSTHi 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 30 p.bevilacquaNov 17, 2009, 5:24am PSTCheck for example, VeraSMART. bsingaraNov 15, 2009, 12:29am PSTcould please someone compare the 3 SHDSL card:<br /><br />Cisco 1-Port G.SHDSL WAN Interface Card (part number WIC-1SHDSL-V3)<br />Cisco 2-Pair G.SHDSL HWIC (HWIC-2SHDSL)<br />Cisco 4-Pair G.SHDSL HWIC (HWIC-4SHDSL)<br /><br />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. <br /><br />Could please light me the best voice WAN solutions between branch office via WAN; 100 users in Dubai and 24 user in Oman.<br /> 30 p.bevilacquaNov 15, 2009, 5:57am PSTYou need to ask telco, if they provide SHDL first. bsingaraNov 15, 2009, 8:21pm PSTTelco is provide SHDL point-to-point connection with bit costly for voip, what is the best design with SHDL or for voip via WAN which would be the best solution, please light me on that.<br /><br />Also please compare the three SHDSL cards, what is the difference between 2 wire and 4 wire. Could Please help us, not reply in one word. p.bevilacquaNov 15, 2009, 10:20pm PSTIt's all explained in the datasheet.<br /><br /><A HREF="javascript:newWin('http://www.cisco.com/en/US/prod/collateral/modules/ps5949/ps7175/product_data_sheet0900aecd80581fa0.html')">http://www.cisco.com/en/US/prod/collateral/modules/ps5949/ps7175/product_data_sheet0900aecd80581fa0.html</A><br /><br />There is no "best solution" all depends by available telco services, and how much you can spend. bsingaraNov 16, 2009, 8:53pm PSTThanks for the reply p.bevilacqua, could you please light me more about 2pair and 4 pair SHDSL card, I am not able to understand the difference between the them. p.bevilacquaNov 16, 2009, 9:28pm PSTTwo wires make a pair.<br />More the pairs, more speed or distance.<br />Probably your telco supports one ir two pairs only.<br /><br /> bsingaraNov 17, 2009, 2:23am PSTThanks again for your reply, if my design supports 4 pair, but the teleco support 2 pair, the SHDSL connection will work or not, so the SHDSL card design should be according to teleco supports, could you please suggest me. p.bevilacquaNov 17, 2009, 5:23am PSTYes, the card must be chosen according to telco offering, as mentioned already.<br /><br />Please remember to rate useful posts with the scrollbox below.<br /><br /> devang_etcomNov 15, 2009, 3:21pm PSTWhich one is widely deployed in real world including enterprise or service provider environments?<br /><br />Is SIP is fully supported on Cisco devices or its still under development ?<br /><br />Thanks,<br />Dev 30virvermaNov 15, 2009, 11:08pm PSTDepends on application, the protocol is deployed.<br />Like in cable service providers, the modems and gateways controlled by mgcp but in other side when PBX got involved H323 is deployed<br /><br />Now a days, lot of service provider are facing out the PRI's and jumping to SIP trunk<br />Lot of cost saving,<br /><br />In cisco enterprise side if you want to get full benefits of SIP, you can check the CUBE product.<br /><br />Let me know if you need more info in this product devang_etcomNov 16, 2009, 7:27pm PSTVIRENDER,<br /><br />Thanks for your info. How bout full SIP support from Cisco equipment perspective? Can you refer me to the link or document where I can see the prectical implementation of MGCP and SIP in SP as well as Enterprise deployment? Can you share more on Cable service provider deployment?<br /><br />So mostly when it comes to internal deployment then we will see more SCCP & SIP endpoint (Phone or any conference devices) but when we are talking about outside communication then H.323 and SIP will play big role, specially when you care communicating with third party voice network...<br /><br /><br />regards,<br />Dev<br />virvermaNov 16, 2009, 7:41pm PSTDevang,<br /><br />Give me one day time, <br />will provide some good stuff on cable service provider stuff as well.<br /><br /> devang_etcomNov 16, 2009, 7:42pm PSTNo prob... I will wait... you can unicast me on <A HREF="mailto:devangnp@gmail.com">devangnp@gmail.com</A> if you want... thanks for your help in advance...<br /><br />regards,<br />DevmohammedabdulNov 17, 2009, 2:49am PSTTo viverma,<br /><br /> If you are not posting all details here then plz mail me also on this E-mail id: <A HREF="mailto:mohammedabdulmalik1430@gmail.com">mohammedabdulmalik1430@gmail.com</A>.<br /><br /> Thanks and regards.virvermaNov 17, 2009, 5:17am PSTI will be doing this on netpro just collecting few information, this is good topic of my interest :)makstilciscoNov 17, 2009, 4:28am PSTHi<br />I have two question and I hope someone can help me<br /><br />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?<br /><br /><br />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?<br /><br />We have CUCM 6.1.3<br /> carbonscoringNov 15, 2009, 5:58pm PSTHi 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.<br /><br />thanks in advance, <br /><br />Mike<br /> 30 hythamhadadNov 16, 2009, 2:36am PST<br />Hi Mike,<br /><br />You can use the attendant console feature of the cucm 6 to make use of its call queueing feature<br />No need for the unity connection<br /><br />Thanks j.schulenbergNov 16, 2009, 11:51am PSTThe Attendant Console built into UC Manager is End of Life and will no longer function starting in 8.0, due out early next year.<br /><br />Your options are to use a hunt group (native to UCM); or, another product such as Contact Center Express. p.bevilacquaNov 16, 2009, 1:04pm PSTBut if I'm not wrong, CM hunt-groups do not do call queuing.<br /><br />Another alternative is B-ACD/AA on the GW in H.323 or SIP mode. That is probably unsupported by cisco.<br /><br /> hythamhadadNov 17, 2009, 4:09am PST<br />Hi,<br /><br />Attendant console won't be available from version 7 fresh install<br /><br />But if he upgrade he'll find it after the upgrade to 7<br /><br />Also you can use the business edition of the attendant console after that<br /><br />Also you can check the B/ACD solution advised above<br /><br /><br /><br />Thanks hythamhadadNov 15, 2009, 1:12pm PST<br />Hi,<br /><br />I have this network diagram<br />CUCM7---GK---CME7<br /><br />CUCM & CME are in the same zone to the GK<br />I need to limit the max bandwidth inside the zone to 64k<br /><br />Bu tI found a strange behavior:<br />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<br /><br /><br />But when calling from CME to CCM <br />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<br /><br />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 <br /><br />Also transcoder is configured on the CME in case the CME will need it<br /><br />So why the CME insist in using 128 K for this call <br /><br />Any feed back will be appreciated<br /><br />thanks 30 hythamhadadNov 17, 2009, 4:06am PST<br />Hi,<br /><br />Is there any help in this issue<br /><br /><br />Thanks nasry_shakoorNov 15, 2009, 11:42pm PSThi,<br /><br />My customer wants me to configure password on meet-me conference !!<br /><br />how can we achieve this ??<br /><br />Regards, 30 rob.huffmanNov 16, 2009, 5:50am PSTHi Ahmed,<br /><br />Sadly, there is no built-in method to do this in CUCM. This question does pop up from time to time;<br /> <br /><br />This solution (paraphrased) is presented in the Cisco Press book - Configuring Callmanager and Unity (A Step by Step Guide) <br /><br />You can use a Unity Call Handler (with the same Extension as the Meet-Me DN) <br /><br />Set up the Call Handler to do a Supervised Transfer to Ring the Meet-Me Conference. <br /><br />Under the "Gather Caller Information" section of the Call Handler check "Confirm" and "Ask Callers Name" <br /><br />When callers go through the Unity Call Handler they are asked to record their name, when the caller is forwarded to the Meet-Me Conference the name is played and conference participants are asked to Press 1 to accept the call or Press 2 to reject the call. <br /><br />There is a nice discussion of this setup in this post (Thanks Deji and Robert); <br /><br /><A HREF="javascript:newWin('http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified')">http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified</A> Communications and Video&topic=IP Telephony&CommCmd=MB?cmd=pass_through&location=outline@^1@@.2cbfa8c4 <br /><br /><br /><br />Hope this helps! <br />Rob <br /><br /> maj@lytzen.dkNov 17, 2009, 2:25am PSTHi Ahmed<br /><br />I have previusly made a script in CCX that solves your problem (almost).<br /><br />The inviter calls a CRS-RP and the system identifies that the call is a internal call, and look up in a database, to see if the conference is active. if the conference is not active, the inviter is asked to enter pincode, and to confirm pincode. Next the inviter is asked for how long the meeting should be active. after this the inviter is told to hang up and initiate Meetme to number XX.<br />Inviter hangs up, and activate Meetme.<br /><br />Participants call in to same CRS-RP as inviter, but now the conference is active, so the participant is asked to enter pincode. If correct the participant is transfered to the meetme number XX.<br /><br />The CRS-RP is reachabel from external, but the meetme numbers are only reachabel from internal.<br /><br />All conferences are in a database, so you can see how often it is used.<br /><br />Hope this could be possible way. <br /><br />Downside is that if you know the number of the meetme, you can bypass the security check.<br /><br />Regards<br /><br />Martin Abildgaard bsingaraNov 17, 2009, 2:26am PSTFor 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. mohammed.naviwalaOct 27, 2009, 5:04am PSTHi there<br /><br />Is this possible??? <br /><br />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. <br /><br />is it possible?? 30 jameshaOct 27, 2009, 11:44am PSTPlease provide details of what system you are using (CallManager or CallManager Express), the version number and the protocol running on the gateway. mohammed.naviwalaNov 10, 2009, 10:59am PSTCall manager 7.0<br /><br />VG224 on SCCP.. i want is if i lift the handset on the analog phone i can dial the number but if i do not dial any number for 6 seconds then it should auto dial to the operator mohammed.naviwalaNov 17, 2009, 2:19am PSTanybody???????????? dwagner@max-boegl.deNov 17, 2009, 1:29am PSTHey,<br />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? jeffshen1215Oct 26, 2009, 9:26am PST<br />3 sites:<br />Site A: Cisco CM 4.2 (212-555-75XX)<br />Site B: Nortel 81C (212-555-79XX)<br />Site C: AVAYA (305-666-45XX)<br /><br />History:<br />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)<br />Project:<br />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.<br />When Site A user dials 9-1-305-666-45XX, CM will route the call via voice gateway to site C. (Done)<br />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)<br />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)<br />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)<br /><br />Troubleshooting:<br />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.<br />Question:<br />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?<br />I appreciate your help!<br /> 30 gogascaOct 26, 2009, 12:05pm PSTFrom Avaya connection to CUCM, create an special partition only assigned to incoming calling search space in Avaya GW in CUCM, this partion will be used ina translation pattern or RP to destination karthikeyan5.pNov 17, 2009, 12:44am PSTHi,<br /><br />I have some queries on the below setup.<br /><br />Current Setup:<br />Site A : CUCM 7.X <br /> Nodes : 1, Publisher - 1 No.<br /> 2, Subscriber - 1 No.<br /> 3, MPX - 1 No.<br /> 4, Unity - 1 No.<br /><br /> DID range : 6742XXXX<br /><br />Site B : Nortel 81 C <br /> DID range : 6616XXXX<br /><br />Connectivity between 2 sites Via Tie PRI for voice.<br /><br />Data connectivity is avaialble between Site A & B.<br /><br />Requirement: TO make Site B as DR for limited Cisco IP Phones users working in Site A.<br /><br />Proposed Scenario:<br />To implement a CUCM 7.X as subscriber in Site B and register the same with CUCM publisher in Site A via Data link.<br /><br />TO registered limited Cisco IP phones in CUCM- SUb available in site B.<br /><br />If CUCM Pub in Site A goes down this subscriber should registered with Nortel 81C and calls should be diverted via this network. (Priority Incoming calls)<br /><br />Query: <br /><br />1, How could I integrate Nortel 81 C with CUCM-SUB in Site B.<br />2, How could I configure Nortel 81 C as secondary in CUCM-Sub in site B in case of CUCM-Pub down in Site A.<br />3, How Could I route particular DID hunt Pilot number configured in CUCM-Pub Site A to CUCM-Sub Site B registered in Nortel 81 C in case of CUCM-Pub down in site A.prakasharNov 3, 2009, 9:20am PSTHi,<br />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.<br /><br />-prakash 30virvermaNov 5, 2009, 8:03pm PSTYou need to backup them manually,<br /><br />The version of CAR that runs on Cisco Unified Communications Manager 6.1(2) <br />does not retain CDRs older than 180 days in the CAR database. If you <br />migrate records older than 180 days, the system deletes them immediately <br />after you upgrade. The Cisco Unified Communications Manager installation <br />program limits the time in which CAR records are migrated from the DMA <br />TAR file to the CAR database on the upgraded system. The installer <br />migrates approximately 100,000 to 150,000 of the oldest individual CAR <br />records within the time limit.<br /><br />Any records that cannot be exported during the specified time will not <br />be migrated."<br /><br /> digisridharNov 16, 2009, 10:04pm PSTHi,<br />How do I backup CDR manually<br /><br />-Sridhar pedroferrerasMay 15, 2009, 12:31pm PSTHi 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.<br />Tahnks.<br /><br /><b>Attachment Keywords : </b> <br />1) Backup without cdr.txt - Backup with cdr.txt<br />2) Backup with cdr.txt<br /> Backup without cdr.txt116186text/plaintext361605/15/2014noBackup with cdr.txt116187text/plaintext413605/15/2014no30vhilarioMay 15, 2009, 4:46pm PSTPedro:<br /><br />If you haven´t purged the cdr database yet, then you will have this kind of issues when backing up the callmanager with cdr database.<br /><br />I will suggest you to manually purge cdr database, I try to backup with bars again, also have you downloaded the latest BARS version for callmanager 4.2 ?<br /><br />Please let me know how everything goes.<br /><br />Also I will appreciate if you could please upload the backup logs pedroferrerasMay 22, 2009, 5:14am PSTAs soon as i purged the cdr database, the backup cpmpleted succesfully.<br /><br />Thank you. digisridharNov 16, 2009, 10:02pm PSTHi,<br />If I purge I will end up losing CDR data. How can I backup this CDR data before purge.<br /><br />-Sridhar littledavewhiteOct 13, 2008, 10:22am PSTAnybody 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? 30kenta_wataiNov 16, 2009, 8:14pm PSTIn the real time monitoring tool, check the ip phone services log4j files.<br /><br />I was able to get it going by an end user account that had devices associated with that account. You might need to be a ccm end user as well. goodwinscottNov 16, 2009, 5:57pm PSTDoes 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?<br /><br />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.<br /><br />FYI, we have CallXpress version 6.5 and Callmanger 4.3. Thanks! thomasmphil123Nov 16, 2009, 4:29pm PSTHi All,<br /><br />Need help.<br /><br />Calls to Meeting Place conference that present over PSTN lines are unable to attend conferences.<br /><br />Callers report the system is doubling the digits sent causing Meeting Place to deny access to the meeting.<br /><br />Please help me on this issue. Mojo1Nov 16, 2009, 3:23pm PSTI 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 ?<br /><br />Thanks ')