Cisco NetPro -

Media Master of unity.

Fri, 3 Jul 2009 14:51:01 PST

Hi All, Media Master tool bar in Unity 4.2 is launched by explorer correctly with correct JAVA plug in. I can record and play the voice but I cannot paste any audio file to the call handler. whenever I try to paste a file I get the following error" unable to paste the audio". Thanks, Mo

issue with ISDN inbound call to IP phones using H.323 GW(Codec problem)

Fri, 3 Jul 2009 14:47:11 PST

Hi I have a H.323 gateway configured with to route calls to CCM using G.711 codecs ( codec configuration within dial-peers config ). When PSTN inbound calls are for IP phones configured with locations stating to use G.729 with other ones, the call cannot be completed. is there anyway to keep using G.711 between the gateway and all others IP devices, and use G.729 between phones belonging to different locations (not using the gateway)? best regards,

Forwarding PSTN inbound call to CTI ports (for operator console)

Fri, 3 Jul 2009 14:35:53 PST

Hi all, I have a H.323 gateway that sends calls to CCM, on its turn , after a translation pattern , th call is forwarded to a unity Conn. Callhandler ( Voicemail Pilot Num) and according to the caller inout digits, the call maybe forwarded to Route Points and then CTI ports , used by an operator console. My problem is I get strange behaviour on the operator console, I got always the calls in the same queue although forwarded to the right number (we get in the right queue if we calle from internal DN), I suppose that is due to codec negociation since CTI ports work only with G.711. What happens when the call get tranferred in term of codec negoiciation ? As the call should use G.729 for the first hop( as specified in a voice class) , is there any further negociation at each hop or the first choice is kept til reaching the final destination ( may be I'm telling silly things ) ? thanks in advance for your help. best regards,

PSTN Incoming Calls cannot be transfered to IP Phones

Fri, 3 Jul 2009 13:34:20 PST

Hi, I do have a Cisco 2650XM with 2 E1s, one for incoming and the other for outgoing calls towards PSTN. When I receive a phone call from the PSTN that reaches the Attendant Console Pilot Number, the Operator can talk with the caller. Once the Operator set the HOLD key on the IP Phone, 1st) there is no MOH stream towards the PSTN, and if the operator would like to UNHOLD the phone call, there is silence on both sides. when I see the codec displayed on the IP Phone, it is g.729. Please help me to understand if this is a mismatch codec issue or something related to rounting? I am using CCM 4.1.3 Thanks in advance for your Help

Off ramp Fax

Fri, 3 Jul 2009 13:05:24 PST

Hi All, Recently i had one POC for FAX over IP using 2821 + E1 PRI. Both On ramp and off ramp fax are working fine , but for off ramp fax with normal TIFF attached file its not transmitting to the other end . I have gone through the docs and came to know Gateway will only support TIFF-F files for off ramp fax. So any applications or converter TIFF-F conversion. Thanks and regards

CUBE Carrier Issue

Fri, 3 Jul 2009 12:56:43 PST

Cutover a customer to a SIP trunk (CUBE) and everything is working fine. You can make outbound calls ok, inbound calls ok. BUT for some reason when a Verizon customer that is local to the area calls this number they will get a busy signal. All the other carriers work ok, Sprint, AT&T, Cricket etc. I tried calling from a remote Verizon area and it works fine...it seems to be only this Verizon carrier in that town. I did both a "debug voice cc inout" and "debug ccsip message". I can see the call hit the gateway fine and negotiate at g.729. The error I am getting from the CUBE router is "SIP/2.0 400 Bad Request - 'Malformed/Missing FROM: field". Does anyone know what this means? I tried sending a call with 0000000 as the calling party but it rang through correctly. Below is the debug v=0 o=CiscoSystemsSIP-GW-UserAgent 5572 7029 IN IP4 10.200.131.42 s=SIP Call c=IN IP4 10.200.131.42 t=0 0 m=audio 17136 RTP/AVP 0 18 c=IN IP4 10.200.131.42 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no Jun 29 16:26:55: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 400 Bad Request - 'Malformed/Missing FROM: field' Reason: Q.850;cause=100 Date: Mon, 29 Jun 2009 21:26:55 GMT From: "IS TECH,SUPPORT" ;tag=48C96BD4-D5E Allow-Events: presence Content-Length: 0 To: ;tag=298619248 Call-ID: 6771D943-642A11DE-9969B8BB-EF9CC056@10.200.131.42 Via: SIP/2.0/UDP 10.200.131.42:5060;branch=z9hG4bK6B91408 CSeq: 101 INVITE

Deleting 7921 / 7925 Ring Tones From Cluster

Fri, 3 Jul 2009 12:50:57 PST

A previous voip engineer uploaded some ring tones that management now wants removed. If I remove those wav files from the TFTP path of all of my CCM servers in my cluster will phones currently configured with one of the deleted ring tones simply default to another ring-tone?

Attendant Console and 10-digit Intersite Dialing

Fri, 3 Jul 2009 12:26:45 PST

Hello, I have 10-digit intersite dialing with 4-digit intrasite dialing setup with translation patterns to accomadate overlapping DN's. In order to get line state working with AC, I have application dial-rules configured to strip the first 6-digit's. This allows the line state to work but breaks AC's ability to call intersite. I know that the overlapping digits will cause problems with the line state, but is there a way to have line state with 4-digits and dialing with 10-digits in AC? Thanks,

SRST over CCME

Fri, 3 Jul 2009 10:16:30 PST

Hi NetPro I can use or configure a gateway whit SRST using a CCME whit call control? I have a two site one whit 200 user ideal for use CCME, the other site have 20 user. My ideal is configure CCME whit one call control and gateway whit SRST working whit CCME. Regards.

CCM6 - how to remove Message softkey from 7911

Fri, 3 Jul 2009 10:11:32 PST

Hi all, When configuring softkey layout in CCM6, I found that there was no Message key in the layout, but with 7911 phone, the key always presents. How can I remove this key as our system has no voice mail! Thanks, hoanghiep

Issues calling out from MPE

Fri, 3 Jul 2009 09:03:17 PST

Hi, we have issues calling out from MeetingPlace Express to landlines and mobiles. Extensions registered to the CCM have no issues. This is a strange one as in the past when a call out failed and the meeting was ended the MPE succesfully called the mobile to advise the meeting has ended. At other times when calling out to a mobile, it did not ring the mobile but left a message on the voicemail. Any help on this would be appreciated. Regards, Darren.

Convert Progress to Disconnect ISDN Q931 Message

Fri, 3 Jul 2009 08:28:27 PST

Hi, currently I have a problem with my QSIG ECMA CUCM <-> siemens connection. PSTN E1 PRI<->H323<->CUCM<->MGCP QSIG ECMA E1 PRI<->SIEMENS the siemens is not delievering a disconnect code, instead it delivers a progress message with the following q931 debug: Jun 16 15:38:46.460: ISDN Se0/1/1:15 Q931: RX <- PROGRESS pd = 8 callref = 0xBC34 Cause i = 0x8091 - User busy disc-pi-off is for sure not the problem, because I tried all possible configurations. The CUCM service parameter does now state the following: Convert Progress to Disconnect for User Side PRI EURO: This parameter determines whether to convert a Progress request that contains a Call Clearing cause value to a Disconnect message. Valid values specify True (convert the Progress request to a Disconnect message) or False (do not convert the Progress request). This parameter is only applicable to the USER side of a PRI EURO interface during the call establishment phase. But this seems not to work, is this because I have a QISG. Does this paramter need a CUCM restart? I have alredy done a mgcp reset (no mgcp/mgcp). Has anyone a clue, what else could help? The problem is, that the caller hers ringback all the time. Would it be possible to play busy back instead of ringback in that case? Currently I’m also not sure if the Siemens could be reconfigured so that it sends a disconnect. any idea is very much appreciated Cheers

[help needed] 7970 CUCM 4.1(3)SR7 Firmware 8.4.1SR1 Issue

Fri, 3 Jul 2009 07:53:50 PST

Dear all, we have upgraded our 3000+ 7970's Phones to the latest 8.4.1SR1 Firmware. Since this time we have had multiple complaints of Users that the screen is now very difficult to read. I have attached 2 screenshot's were hopefully the problem is clearer to see. It seems the font now has shadow and "bleeds" into the background picture. The example Photo with the large glass building is photographed in a way that the sky is visible. In case of a Voice Mail as the envelope is white it is very hard to see. The 2nd attached photo also shows the now larger Dialog box when dialling 12. The white in the screenshot is not true to the actual white that is displayed on the Phone screen. Changing the brightness or the viewing angle does not really sort the problem. The 2nd issue with this firmware is that all headset's used within the company now have to be readjusted in order to not have an echo. This was not the case with the previousely used Firmware 8.3.x Any hints ? Attachment Keywords : 1) Cisco_Firmware.png 2) Cisco_Firmware_2.png

CUCM 6.1.2 - Multi Level Administration

Fri, 3 Jul 2009 07:31:12 PST

Hi, I have a system where we'd like to grant various levels of admin rights to make changes. Currently - I can't find a way of allowing someone to subscribe phones and device profiles to services without giving them full super level access which just isn't possible. Does anyone have a solution to this or is it just an error on my system? Cheers Dave

Part Numbers CUCM

Fri, 3 Jul 2009 07:25:36 PST

Hi NetPro I have this part number CM6.1U4-K9-7825= how is this? Only the media, media and licence, or media, licence and hardware? Regards.

Unity Connection 7 - Media Master Access denied error

Fri, 3 Jul 2009 06:53:27 PST

Hi, Running UC7 and when trying to record a new call handler via the Media Master through a phone handset - I get the following error message: Server Error: The server 192.168.10.10 reports the following: Code 23 Description: Access Denied Anyone else seen this or know how to get round it? I can record the messge via Sound recorder for the time being but I'd like to get this working. Regards Dave

Caller Input + Unity + Forwarding

Fri, 3 Jul 2009 06:48:24 PST

I am attempting to do the following: Someone calls (internal) x8000, get's voicemail(Unity) and the option to transfer to someone else. Option 1 is selected, which transfers to x8001. x8001 is set to forward all calls to an external 800 #. The problem is, when option 1 is selected, it does forward to the x8001 but then goes to a busy signal. If I dial x8001, it does transfer correctly to the external 800 # Any ideas are welcome.

ccmuser page session timeout

Fri, 3 Jul 2009 06:25:04 PST

Hi Is there a way to increase the timeout of the ccmuser page in CUCM 7.x? Cheers Peter

Callback option

Fri, 3 Jul 2009 06:14:54 PST

hi friends, hiw the callback option configure in the ccme. i got one pdf ,but in that method i cannot find callback option in my ios command my version is Version 12.4(15)T. send how to configure also. thank cyril

Cisco Unity Server integration with Microsoft Outlook 2007

Fri, 3 Jul 2009 06:02:10 PST

Hi, Can someone guide me with the integration steps between Cisco unity sever 7.0 & Microsoft Outlook 2007....I have AD 2008...

7921 used with ISDN (1721 router)

Fri, 3 Jul 2009 05:22:12 PST

Im experimenting with a 1721 router. Does anyone know if i can make the 7921/7920 wireless ip phones work to use Voip over isdn? If so could anyone point me in the right direction to making it all work.

tdm gateway modem passthrough

Fri, 3 Jul 2009 05:20:49 PST

Hi! The problem is the following: the customer had a TDM PBX connected to PSTN through E1 PRI. After implementing IP-telephony we installed Cisco3825 with 2 PRIs on vwics (1st facing PSTN, 2nd facing local PBX) to host applications and allow PSTN access from voip peers. The problem is that modem calls from modems attached to PBX towards PSTN have become unstable, loosing connection after ~1 min. Gateway sees it as normal clearing on local modem side. In this case gateway 3825 simply switches calls from one PRI to the other, but it brings in this problem. What could be done to solve it? There is no modem detection implemented anywhere now and I don't know why adding one voice switch could lead to that. Any suggestions would be appreciated.

Problem change language 7911 phone

Fri, 3 Jul 2009 05:20:08 PST

I have to change the locale languages on my phone (7911,7945) with cme 4.1. There isn't any problem with 7945 phone, but I'm not able to change the languages on 7911 phone. This is my configuration: tftp-server flash:apps11.8-2-2ES1.sbn tftp-server flash:cnu11.8-2-2ES1.sbn tftp-server flash:cvm11sccp.8-2-2ES1.sbn tftp-server flash:dsp11.8-2-2ES1.sbn tftp-server flash:jar11sccp.8-2-2ES1.sbn tftp-server flash:SCCP11.8-2-2SR1S.loads tftp-server flash:term06.default.loads tftp-server flash:term11.default.loads tftp-server flash:apps45.8-3-1-22.sbn tftp-server flash:cnu45.8-3-1-22.sbn tftp-server flash:cvm45sccp.8-3-1-22.sbn tftp-server flash:dsp45.8-3-1-22.sbn tftp-server flash:jar45sccp.8-3-1-22.sbn tftp-server flash:SCCP45.8-3-2S.loads tftp-server flash:term45.default.loads tftp-server flash:term65.default.loads tftp-server flash:music-on-hold.au tftp-server flash:/it-be-sccp.jar alias Italian_Italy/be-sccp.jar tftp-server flash:/it-tc-sccp.jar alias Italian_Italy/tc-sccp.jar tftp-server flash:/it-ipc-sccp.jar alias Italian_Italy/ipc-sccp.jar tftp-server flash:/g3-tones.xml alias Italy/g3-tones.xml ! telephony-service no auto-reg-ephone load 7945 SCCP45.8-3-2S load 7911 SCCP11.8-2-2SR1S max-ephones 58 max-dn 116 ip source-address 192.168.11.253 port 2000 timeouts interdigit 3 user-locale IT network-locale IT time-zone 23 time-format 24 date-format dd-mm-yy max-conferences 4 gain -6 call-forward pattern .T call-forward pattern 0T moh music-on-hold.au dn-webedit time-webedit transfer-system full-consult dss transfer-pattern 0T secondary-dialtone 0 create cnf-files version-stamp Jan 01 2002 00:00:00 ! ephone-dn 1 dual-line number 122 pickup-group 1 label Sb name Sb huntstop channel ! ! ephone-dn 2 dual-line number 123 pickup-group 1 label Sa name Sa huntstop channel ! ephone 1 mac-address 0024.97A8.E4A6 ephone-template 1 type 7911 button 1:1 ! ephone 2 mac-address 0024.97A8.E0EB ephone-template 1 type 7945 button 1:2 !

Error Msg not coming

Fri, 3 Jul 2009 04:29:59 PST

hi all, guys i faced a strange prob.i am not able to hear error msg " Your call can not be completed as dial plz confirm your directory and call again" when i call to number which is in different parition ... if i dial such a number then i didnt hear msg and after some seconds call disconnected automatically. i have ccm 6.1 version does any one have idea from where this msg use to come from ccm ??? Regards,

integration b/w CME and active directory

Fri, 3 Jul 2009 03:40:51 PST

hi, Is it possible to integrate CME with Active directory to get username, password and corporate directory information. Regards Naresh

Cisco Click to call

Fri, 3 Jul 2009 03:40:24 PST

This little widget is a great tool with our new VoIP UCM 7.0 except for one thing. I can't get it to work with Outlook 2007. It works fine in IE, Excel and Word. I just right click on the number and place a call. If I go to the contact name in Outlook and right click on the number then go to call Contact... it uses the default Outlook protocol to place the call and not Click to call.

tdm gateway modem passthrough

Fri, 3 Jul 2009 02:51:23 PST

CUCM Conference Bridge with XCode ressouces low level voice

Fri, 3 Jul 2009 02:38:24 PST

I use CUCM software CFB and Xcode ressources on 2811 with PVDM2-64 DSP's. Many members join the conferences through an IP Telco, in G729 and they find the voice level too low. For other members on the Head-Quater site (in G711) and members over the WAN (in G729), it seems better. My question: is it possible to add somewhere some gain, like you can do on voice ports ?

Conference calls dropped after 5 to 15 min.

Fri, 3 Jul 2009 02:08:55 PST

Hi all, I am having such an extrange problem. I have a CM with 6.1.3 with a MGCP Gateway 2851. When conferencing with numbers outside the network, after a while, it becomes released with this reason: *Jul 2 16:04:47.037: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x0A39 Cause i = 0x80A9 - Temporary failure I have the CM configured with g711/g722 for intraregion conferencing and g729 for interregion conferencing. The gateway is not configured as conferencing bridge. Must I? Any clue¿? Thanks in advance

VG224 problem

Fri, 3 Jul 2009 01:46:21 PST

Hi I have a problem with VG224. Analog phones which are connected don't work. In configuration vioce-ports disappear. Yesterday VG224 worked fine. I put in attachment sh diag, sh inventory, boot process and startup-configuration. BR, Miroslav Attachment Keywords : 1) sh inventory sh diag.txt 2) vg224-3 boot process.txt - vg224 running config.txt 3) vg224 startup config.txt