Re: [OSL | CCIE_Voice] Problem with Multicast MOH from router flash
The show ccm music will not show you any output for Voip calls , this is used when you call from the PSTN to the BR1 and place the call on hold ( in other word when the DSP get involved as the ccm music command is to make the DSP injecting the MOH stream into the Isdn Call ) so this is normal , I think you Missing the Voice class codec under the incoming Voip Dial-peer on the BR1 router , you need it as the Call will be swapped to G711u when you will place it on hold , make sure you have it , Also what do you hear when you Put the PSTN call on hold ? in this case you can use the show ccm music to see what codec you are using , Ash On Thu, Dec 1, 2011 at 1:24 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: But I am streaming the MoH from the branch router ,so it should not be using a dial-peer ? Or does it still use the dial-peer as it thinks it is streaming from the CUCM server ? From: William Affeldt [mailto:william.affe...@yahoo.com] Sent: 30 November 2011 08:46 PM To: Rynard Coetzee Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with Multicast MOH from router flash Make sure that you configured the codec under the dial peers on BR1. They use g729 by default. Sounds like the dial-peers to call manager from BR1 don't have a voice class codec or are a hard set codec configured. Sent from my iPhone On Nov 30, 2011, at 8:12 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Hi guys I have been struggling with this problem for most of the day now ,and I don`t know what else I can check. The MoH works if I call from HQ branch to BR1 ,but when I call from the PSTN to BR1 I get tone on hold. I have the relevant config on my BR1 router Ccm-manager music-on-hold Telephony service Moh music-on-hold.au Multicast moh 239.1.1.1 port 16384 route 10.10.110.2 (loopback) 10.10.201.1 (Voice Int) I also configured separate DP for MOH server in CUCM with a G711 only region. I have the MOH setting on the audio source ,server and MRG set. Another strange thing that I see is when I call from HQ phone to BR1 phone and I put BR1 phone on hold I hear the MOH stream ,but when I do show ccm-manager music-on-hold it shows that there are 0 active multicast sessions ? Any ideas ,because I have run out … ? Regards Rynard ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Adding second language to CUE
This mean you are installing the Wrong Language files , or you missing on critical file , can you please paste what you have in the FTP directory root ? Ash On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to add a second language to an AIM-CUE. I use the command software install add url ftp://x.x.x.x/xyz.pkg and it seems to run without a problem but when it finishes processing the file, I get the follow message : Language add-ons found on the system (1): Installed SKU Name (version) -- * ENU CUE Voicemail US English (7.0.6) Maximum allowed language add-ons (=1) already installed. You can use software uninstall to remove add-ons. ui_install scripts executed successfully. The issue is if I run Show software licenses , it indicates a max of 2 languages are allowed. CUE# sho software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 2 - Max enabled languages: 2 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Softkey Templates with CME (ephone-template)
Hi, If I want to add one softkey to a certain state of a phone , I have to create a ephone template with all the required softkeys for the state apply to ephone. For that I have to check other default softkeys by making a call from this phone checking softkeys. But this is time consuming. Is there a quicker way to just add only one softkey in CME ? Ken ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Voice Rack
Hi, I have the required devices to study CCIE LAB v.3, it does not match the requirements exactly but it is very good. To save my time I need to upload the initial configuration for each device at the beginning of each lab. I need to build an automated way to do this rather than do everything manually, like IP-Expert rack, anybody can help me to do this and suggest the best solutions for this? Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2: cannot get inbound H.323 trunk calls to CME SIP phone to work G.729 to G.711
Tks Ash! That explains the solution guide where there are no voice classes on thIs trunk, although in trunks to PSTN breakout Vik invariably puts the voice-class in the dial peer. On 1 Dec 2011, at 15:41, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello Anthony , You cannot Transcode call that Hit Dial peer with Voice class codec , it make sense as the router though that he can support Both codecs I hope this clarify the issue you saw Ash On Wed, Nov 30, 2011 at 8:36 PM, Anthony Alba ascanio.al...@gmail.com wrote: Very strange: I can now get both inbound and outbound calls to CME SIP working with transcoder invoked at BR2-RTR. I cannot use voice-class codec 1 under the dial-peer. This surprises me: why would voice class codec hurt the task? voice class codec 1 codec pref 1 g729r8 codec pref 2 g711ulaw If I put this under any of the dial-peers it breaks CME SIP. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2: cannot get inbound H.323 trunk calls to CME SIP phone to work G.729 to G.711
When you have Voice Class Codec for incoming Dial-peer on CUCME, with G711 and G729, the calls to CUE are not transcoded, and the calls are failing. That's because CUE uses only G711 and as the G711 is also found in the Voice Class codec, the call is trying to take G711, whereas CUCM IP Phones are using G729, and the call is failed. You need to hardcode only one codec (G729r8 for instance) for incoming calls on a dial-peer, so that the calls to CUE will be transcded always. On Thu, Dec 1, 2011 at 3:41 PM, Anthony Alba ascanio.al...@gmail.comwrote: Tks Ash! That explains the solution guide where there are no voice classes on thIs trunk, although in trunks to PSTN breakout Vik invariably puts the voice-class in the dial peer. On 1 Dec 2011, at 15:41, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello Anthony , You cannot Transcode call that Hit Dial peer with Voice class codec , it make sense as the router though that he can support Both codecs I hope this clarify the issue you saw Ash On Wed, Nov 30, 2011 at 8:36 PM, Anthony Alba ascanio.al...@gmail.com wrote: Very strange: I can now get both inbound and outbound calls to CME SIP working with transcoder invoked at BR2-RTR. I cannot use voice-class codec 1 under the dial-peer. This surprises me: why would voice class codec hurt the task? voice class codec 1 codec pref 1 g729r8 codec pref 2 g711ulaw If I put this under any of the dial-peers it breaks CME SIP. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [EMEA Cluster Gateway Dial Plan Assistance Munich Germany]
Greetings - I am seeking input on developing a dial plan for a site that has been thrown my way in Munich Germany. I'm new to ISDN ERA and have been using NANP for years. Any input regarding developing a dial plan for Munich Germany, including sample configurations of CUCM and Gateways, would be greatly appreciated or if you can point me to resources ( in the right direction) would be immensely appreciated. Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Adding second language to CUE
Hi Ashraf, See below. Thank you! ftp ls 200 Port command successful 150 Opening data channel for directory list. cue-installer.nm-aim.7.0.1 cue-installer.nm-aim.7.0.6 cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1 cue-vm-full-k9.nm-aim.7.0.1.prt1 cue-vm-full-k9.nm-aim.7.0.6.prt1 cue-vm-installer-k9.nm-aim.7.0.1.prt1 cue-vm-installer-k9.nm-aim.7.0.6.prt1 cue-vm-k9.nm-aim.7.0.1.pkg cue-vm-k9.nm-aim.7.0.1.zip cue-vm-k9.nm-aim.7.0.6.pkg cue-vm-k9.nm-aim.7.0.6.zip cue-vm-k9.nmx.7.1.2.zip cue-vm-langpack.nm-aim.7.0.1.pkg cue-vm-langpack.nm-aim.7.0.6.pkg cue-vm-license_12mbx_ccm_7.0.1.pkg cue-vm-license_12mbx_ccm_7.0.6.pkg cue-vm-license_12mbx_cme_7.0.1.pkg cue-vm-license_12mbx_cme_7.0.6.pkg On Thu, Dec 1, 2011 at 2:57 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: This mean you are installing the Wrong Language files , or you missing on critical file , can you please paste what you have in the FTP directory root ? Ash On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to add a second language to an AIM-CUE. I use the command software install add url ftp://x.x.x.x/xyz.pkg and it seems to run without a problem but when it finishes processing the file, I get the follow message : Language add-ons found on the system (1): Installed SKUName (version) -- * ENU CUE Voicemail US English (7.0.6) Maximum allowed language add-ons (=1) already installed. You can use software uninstall to remove add-ons. ui_install scripts executed successfully. The issue is if I run Show software licenses , it indicates a max of 2 languages are allowed. CUE# sho software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 2 - Max enabled languages: 2 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Need help understanding this behavior
Priyank, Question is WHY MTP? On Thu, Dec 1, 2011 at 4:33 AM, Priyank Kiran priyank.ki...@gmail.comwrote: Thanks, now do you guys know if this is going to be addressed in later revisions of CUCM (8.x) OR changing from H323 to SIP would help? OR may be its just the way MTP behaves and you just can't support MMoH with MTP involved? Priyank On Wed, Nov 30, 2011 at 5:14 PM, Mohammed Al Baqari baqari.voic...@gmail.com wrote: Guys,b ** ** MMoH isn’t supported as per Cisco when MTP is invoked. Instead you will hear ToH. ** ** “The following restriction exists for multicast music on hold (MOH) when a media termination point (MTP) is invoked. When an MTP resource gets invoked in a call leg at a site that is using multicast MOH, the caller receives silence instead of music on hold. To avoid this scenario, configure unicast MOH or Tone on Hold instead of multicast MOH.” ** ** http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsmoh.html ** ** Regards, Mohammed Al Baqari ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Robert Thomas *Sent:* Saturday, November 26, 2011 10:43 AM *To:* Divin Mathew John *Cc:* ccie_voice@onlinestudylist.com; Priyank Kiran *Subject:* Re: [OSL | CCIE_Voice] Need help understanding this behavior** ** ** ** Divin is right. MTP doesnt support connecting to a multicast stream, so it would make sense CUCM converts it to a unicast stream. On Fri, Nov 25, 2011 at 11:32 PM, Divin Mathew John divinj...@gmail.com wrote: I think this is normal behaviour. MULTICAST MOH cannot go thru MTP. as in the MTP cannot re-originate the stream. CUCM uses Unicast MOH instead. At least in the lab scenario, try not to check MTP box. ** ** On Fri, Nov 25, 2011 at 1:58 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: I Got real case for the same issue but longtime ago , cannot remember 100% what i have done on it can you please collect ccm sdi sdl and IPVMS detailed traces for the both calls ? I will may take a look when i will have Chance to , but i also need to know why you have enabled the MTP first place and what type of MTP we have IOS or CCM MTP ? Ash On Thu, Nov 24, 2011 at 4:42 PM, Priyank Kiran priyank.ki...@gmail.com wrote: It does On Thursday, November 24, 2011, Mohd Baqari baqari.voic...@gmail.com wrote: The MRGL of MTP should have MoH multicast. Regards, Mohammed Al Baqari Sent from my iPhone On Nov 25, 2011, at 1:57 AM, Priyank Kiran priyank.ki...@gmail.com wrote: No it's not, have 2 MRGLs 1) MRGL attached to DP of gateway and MTP are same mrgl-siteXX mrg-moh + mrg-mtp-siteXX 2) MRGL attached to DP of MoH server mrgl-moh mrg-moh Would like to point out that I only have 1 muticast source and server in my cluster which has been bound to mrg-moh. On Thu, Nov 24, 2011 at 4:44 PM, Mohd Baqari baqari.voic...@gmail.com wrote: What is the MRGL assigned to the device pool of your MTP. Is it the same MoH multicast MRGL. Regards, Mohammed Al Baqari Sent from my iPhone On Nov 24, 2011, at 9:50 PM, Priyank Kiran priyank.ki...@gmail.com wrote: Experts, Need help understanding the following behavior conceptually - Have the subscriber as dedicated MOH multicast server incrementing on port with default address 239.1.1.1 port 16384 Remote H323 gateway, with local music-on-hold wav file spoofing the above source address. This works as expected when put on HOLD and I see all the right output via show ccm-manager music-on-hold and debug ccm-manager music events and show perf query class However, when I check the Media Termination Point Required box on the gateway page in CUCM - I no longer see it sourcing off of the local router flash and it now becomes a unicast stream sourcing off of the Subscriber which I can see from the show perf query class command. Couple questions I have is 1) What forces it to go unicast when you check the MTP required box? 2) Can you still have multicast music-on-hold stream off the local router flash with MTP required check ON? Thanks, Priyank ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you
[OSL | CCIE_Voice] Automated Reply Re: CCIE_Voice Digest, Vol 70, Issue 4
This is an automated reply to your message CCIE_Voice Digest, Vol 70, Issue 4 sent to stewart.mcfarl...@provista-uk.com. Dear ccie_voice-requ...@onlinestudylist.com Thank you for your email. Please note that due to unforseen circumstances I will be out of the office for the rest of this week (w/c 17/10/11). If you require any assistance please contact the provista office on 08456424642. For technical assistance please contact serviced...@provista-uk.com. Thanks Stewart ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.
Use the command debug voice ccapi inout. H323 debugs won't show in this case. Regards, Mohammed Al Baqari Sent from my iPhone On Dec 1, 2011, at 6:12 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to setup a call from HQ CUCM via GK-Trunk to a Remote Gk Zone. I have the Gatekeeper configured with OutVia for the remote zone referencing a CUBE on the HQ router. I didn't realize (but it makes sense now) that with Wait for H.245 unchecked on on the CUCM trunk, the call setup goes to the GK/CUBE as g.711. This obviously causes a problem when the CUBE (by default) tries to create the outgoing call leg to the remote zone using G.729. I don't have an XCoder available to CUBE at this point. My problem is that I can't see the codec mismatch failure in debug cch323 h225 or debug cch323 h245. (If it's in there , I'm not seeing :) ) Can someone help me understand if the failure is noted in either of these debugs Or Point me towards a debug that would show the codec mismatch problem? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Need Help with MoH
Great Post. Thanks you so much. I will test it in my LAB tomorrow and will post the results as well. Initially my issue was caused by the incorrect dspfarm profile for Xcoder. I missed to put the G729r8 codec over there, so the Xcoder did not work, but as soon as I have added that codec to Xcoder the MoH was transcoded :) But the BUG you have sent, is very intestring, so I will also test it tomorrow. On Thu, Dec 1, 2011 at 7:57 PM, Mohammed Al Baqari baqari.voic...@gmail.com wrote: I have tested this and got the result successful using MTP + XCODE. I used exactly same scenario. Here is the output. ** ** HQ#sh call leg act su Gid Lid Elog A/O FAX Tsec Codec typePeer Address IP Rip:udp G0 L 83 N ORG T10g729r8 VOIPP 142.2.66.254:17734 G0 L 85 N ORG T4 g729r8 VOIPP 142.2.64.254:17672 G0 L 86 N ORG T4 g729r8 VOIPP 142.2.64.254:17502 G0 L 88 N ORG T4 g711ulawVOIPP 0.0.0.0:0 ** ** The interesting part is this. It wasn’t working without MTP (either I am using g711 on MOH which needs XCODE or even if I change MOH to G729 which doesn’t require XCODE). Finally I found that I am hitting this bug. So it’s a must to use MTP over SIP trunk for MOH to work either with XCODE or not. ** ** *CSCso85618 Bug Details * ** ** Top of Form *No Audio when Call is put on hold by remote party over Sip Trunk * *Symptoms*: 1. The MOH does not work between CM and CME. 2. There is no audio on the CME endpoint when the remote CCM party resumes the call on hold, conferences, or transfers with another CCM endpoint (scenario: CME - CUBE - CCM). *Conditions*: Symptom 1 is observed if the phone registered to the CME is put on hold by the CM, then the CME phone does not hear the MOH. Symptom 2 is observed if the CCM endpoint does a conference, hold, or transfer. Workaround: Use an MTP. Regards, Mohammed Al Baqari ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123 datucha123 *Sent:* Wednesday, November 30, 2011 10:59 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Need Help with MoH ** ** Hello, I am facing a Problem with MoH over the SIP/H323 Trunk with MTP. I know, that the multicast MoH is not supported when the MTP is check in the H323 gateway, but I have a problem with Unicast MoH with transcoding.* *** But I have the following configuration. Only G711U is activated for MoH (default). I have a SIP Trunk towards CUCME, this SIP Trunk is in G729 Region, so that this Region is using G729 to CUCM IP Phones, and to MoH regions as well: So I have HQ_Region - G711 Inside and G729 to SIP_Trunk_Region. SIP_Trunk_Region - G711 inside and G729 to HQ_Region. So the SIP Trunk is assigned the SIP_Trunk_Region (Through DP) and the IP Phones along with the MoH Server are assigned the HQ_Region. Also I have an G729 MTP on a Router, which is assigned the MTP_Region (Which is using G729 to all other Regions). Transcoder is assigne the HQ_Region. So when I call the SIP Trunk (CUCME IP Phones) the call is successful and negotiated codec is G729, and MTP is also allocated, as I have checked the MTP Required Checkbox. But when I press the Hold key on the CUCM IP Phone, the CUCME IP Phone hears a Beeps instead of MoH Music. I cannot get how to make transcoder to transcode the MoH from G711 to G729, so that the MTP will be also used on the SIP Trunk. BTW, when I remove the MTP Required from the SIP Trunk, the MoH is transcoded, and the remote IP Phone (CUCME) can hear the music. But I want to have an MTP checked as well, and in this case the MoH music does not play. (Only Beeps). Please help me. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Adding second language to CUE
Hello , why you have tow packages in the root directory ? you have to have the full package of 7.0.6 and the lang pack of GB 7.0.6 ONLY on the root directory , run the installation again and see how it will go Ash On Thu, Dec 1, 2011 at 8:02 AM, ccielabrat ccielab...@gmail.com wrote: Hi Ashraf, See below. Thank you! ftp ls 200 Port command successful 150 Opening data channel for directory list. cue-installer.nm-aim.7.0.1 cue-installer.nm-aim.7.0.6 cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1 cue-vm-full-k9.nm-aim.7.0.1.prt1 cue-vm-full-k9.nm-aim.7.0.6.prt1 cue-vm-installer-k9.nm-aim.7.0.1.prt1 cue-vm-installer-k9.nm-aim.7.0.6.prt1 cue-vm-k9.nm-aim.7.0.1.pkg cue-vm-k9.nm-aim.7.0.1.zip cue-vm-k9.nm-aim.7.0.6.pkg cue-vm-k9.nm-aim.7.0.6.zip cue-vm-k9.nmx.7.1.2.zip cue-vm-langpack.nm-aim.7.0.1.pkg cue-vm-langpack.nm-aim.7.0.6.pkg cue-vm-license_12mbx_ccm_7.0.1.pkg cue-vm-license_12mbx_ccm_7.0.6.pkg cue-vm-license_12mbx_cme_7.0.1.pkg cue-vm-license_12mbx_cme_7.0.6.pkg On Thu, Dec 1, 2011 at 2:57 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: This mean you are installing the Wrong Language files , or you missing on critical file , can you please paste what you have in the FTP directory root ? Ash On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to add a second language to an AIM-CUE. I use the command software install add url ftp://x.x.x.x/xyz.pkg and it seems to run without a problem but when it finishes processing the file, I get the follow message : Language add-ons found on the system (1): Installed SKU Name (version) -- * ENU CUE Voicemail US English (7.0.6) Maximum allowed language add-ons (=1) already installed. You can use software uninstall to remove add-ons. ui_install scripts executed successfully. The issue is if I run Show software licenses , it indicates a max of 2 languages are allowed. CUE# sho software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 2 - Max enabled languages: 2 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] [EMEA Cluster Gateway Dial Plan Assistance Munich Germany]
Thanks Ash - Yeah I'll speak to BT thought this was going to be a tough one. -Original Message- From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com] Sent: Thursday, December 01, 2011 11:39 AM To: Sears, Michael (msears) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [EMEA Cluster Gateway Dial Plan Assistance Munich Germany] Germany Dial-plan is the most complicated dial-plan in the world because its not organized as the NANP or UK Dial plans , they don't have a fixed pattern in the access codes or length , you have to speak to someone from Germany who know about the telecom there or maybe the German telecom itself can give you guide about that , Ash On Thu, Dec 1, 2011 at 7:53 AM, michael.se...@compucom.com wrote: Greetings - I am seeking input on developing a dial plan for a site that has been thrown my way in Munich Germany. I'm new to ISDN ERA and have been using NANP for years. Any input regarding developing a dial plan for Munich Germany, including sample configurations of CUCM and Gateways, would be greatly appreciated or if you can point me to resources ( in the right direction) would be immensely appreciated. Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.
The ccapi debug will show you the cause code which doesn't explain why the call failed , you have to debug the h245 asn1 and check the TCS and see the codecs advertised and received and then you will get the TCS negotiation failure so you can explain that there is codec mismatch Ash On Thu, Dec 1, 2011 at 11:55 AM, Mohd Baqari baqari.voic...@gmail.com wrote: Use the command debug voice ccapi inout. H323 debugs won't show in this case. Regards, Mohammed Al Baqari Sent from my iPhone On Dec 1, 2011, at 6:12 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to setup a call from HQ CUCM via GK-Trunk to a Remote Gk Zone. I have the Gatekeeper configured with OutVia for the remote zone referencing a CUBE on the HQ router. I didn't realize (but it makes sense now) that with Wait for H.245 unchecked on on the CUCM trunk, the call setup goes to the GK/CUBE as g.711. This obviously causes a problem when the CUBE (by default) tries to create the outgoing call leg to the remote zone using G.729. I don't have an XCoder available to CUBE at this point. My problem is that I can't see the codec mismatch failure in debug cch323 h225 or debug cch323 h245. (If it's in there , I'm not seeing :) ) Can someone help me understand if the failure is noted in either of these debugs Or Point me towards a debug that would show the codec mismatch problem? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Adding second language to CUE
oh, ok. I'll give it a try. On Thu, Dec 1, 2011 at 1:44 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello , why you have tow packages in the root directory ? you have to have the full package of 7.0.6 and the lang pack of GB 7.0.6 ONLY on the root directory , run the installation again and see how it will go Ash On Thu, Dec 1, 2011 at 8:02 AM, ccielabrat ccielab...@gmail.com wrote: Hi Ashraf, See below. Thank you! ftp ls 200 Port command successful 150 Opening data channel for directory list. cue-installer.nm-aim.7.0.1 cue-installer.nm-aim.7.0.6 cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1 cue-vm-full-k9.nm-aim.7.0.1.prt1 cue-vm-full-k9.nm-aim.7.0.6.prt1 cue-vm-installer-k9.nm-aim.7.0.1.prt1 cue-vm-installer-k9.nm-aim.7.0.6.prt1 cue-vm-k9.nm-aim.7.0.1.pkg cue-vm-k9.nm-aim.7.0.1.zip cue-vm-k9.nm-aim.7.0.6.pkg cue-vm-k9.nm-aim.7.0.6.zip cue-vm-k9.nmx.7.1.2.zip cue-vm-langpack.nm-aim.7.0.1.pkg cue-vm-langpack.nm-aim.7.0.6.pkg cue-vm-license_12mbx_ccm_7.0.1.pkg cue-vm-license_12mbx_ccm_7.0.6.pkg cue-vm-license_12mbx_cme_7.0.1.pkg cue-vm-license_12mbx_cme_7.0.6.pkg On Thu, Dec 1, 2011 at 2:57 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: This mean you are installing the Wrong Language files , or you missing on critical file , can you please paste what you have in the FTP directory root ? Ash On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to add a second language to an AIM-CUE. I use the command software install add url ftp://x.x.x.x/xyz.pkg and it seems to run without a problem but when it finishes processing the file, I get the follow message : Language add-ons found on the system (1): Installed SKU Name (version) -- * ENU CUE Voicemail US English (7.0.6) Maximum allowed language add-ons (=1) already installed. You can use software uninstall to remove add-ons. ui_install scripts executed successfully. The issue is if I run Show software licenses , it indicates a max of 2 languages are allowed. CUE# sho software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 2 - Max enabled languages: 2 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.
ok, I'm getting to understand this better.
I don't see any mention of a tcs failure though
See the output of debug h245 asn1 below. Where is the indication of a
failure?
Also, I have CUBE running with a Hw transcoder registered locally on HQ
telephony-service.
I would think the CUBE should allocate the xcoder to get around the codec
mismatch.
Output from Debug of H245 ASN1 on HQ/GK/CUBE
Dec 1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0, bytesLeftToDecode
= 97
Dec 1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
Dec 1 20:45:36.806:
Dec 1 20:45:36.806: H245 MSC INCOMING PDU ::=
value MultimediaSystemControlMessage ::= request : terminalCapabilitySet :
{
sequenceNumber 1
protocolIdentifier { 0 0 8 245 0 10 }
multiplexCapability h2250Capability :
{
maximumAudioDelayJitter 60
receiveMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FA
HQ#LSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
transmitMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FALSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
receiveAndTransmitMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FALSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
mcCapability
{
centralizedConferenceMC FALSE
decentralizedConferenceMC FALSE
}
rtcpVideoControlCapability FALSE
mediaPacketizationCapability
{
h261aVideoPacketization FALSE
}
logicalChannelSwitchingCapability FALSE
t120DynamicPortCapability FALSE
}
capabilityTable
{
{
capabilityTableEntryNumber 1
capability receiveAudioCapability : g729wAnnexB : 6
},
{
capabilityTableEntryNumber 2
capability receiveAudioCapability : g729AnnexAwAnnexB : 6
},
{
capabilityTableEntryNumber 3
capability receiveAudioCapability : g729 : 6
},
{
capabilityTableEntryNumber 4
capability receiveAudioCapability : g729AnnexA : 6
},
{
capabilityTableEntryNumber 5
capability receiveAndTransmitUserInputCapability : dtmf : NULL
},
{
capabilityTableEntryNumber 6
capability receiveAndTransmitUserInputCapability : basicString :
NULL
},
{
capabilityTableEntryNumber 44
capability receiveAndTransmitUserInputCapability : hookflash :
NULL
}
}
capabilityDescriptors
{
{
capabilityDescriptorNumber 0
simultaneousCapabilities
{
{
1,
2,
3,
4
},
{
5,
6
},
{
44
}
}
}
}
}
Dec 1 20:45:36.810: h245_decode_one_pdu: H245ASNDecodePdu rc = 0,
bytesLeftToDecode = 0
Dec 1 20:45:36.810: h245_decode_one_pdu: Read Pkt body: more_pdus:0 rc:0
asn_rc:0
HQ#
HQ#
HQ#
HQ#sho deb
H.245:
H.245 ASN1 Messages debugging is on
On Thu, Dec 1, 2011 at 2:00 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
The ccapi debug will show you the cause code which doesn't explain why
the call failed ,
you have to debug the h245 asn1 and check the TCS and see the codecs
advertised and received and then you will get the TCS negotiation
failure so you can explain that there is codec mismatch
Ash
On Thu, Dec 1, 2011 at 11:55 AM, Mohd Baqari baqari.voic...@gmail.com
wrote:
Use the command debug voice ccapi inout. H323 debugs won't show in this
case.
Regards,
Mohammed Al Baqari
Sent from my iPhone
On Dec 1, 2011, at 6:12 PM, ccielabrat ccielab...@gmail.com wrote:
I'm trying to setup a call from HQ
Re: [OSL | CCIE_Voice] CME as SRST won't forward to voicemail using ephone-dn-template
Did you try adding a ephone-dn (octo line) with conference adhoc for cBarge? On Fri, Dec 2, 2011 at 5:24 AM, ccielabrat ccielab...@gmail.com wrote: Hey all, I'm finishing up testing my CME as SRST testing and ran in to a problem where it looks like the phones are getting the ephone-dn-template configured under telephony-service. The phone just rings and rings I'm also wondering if the DND button should automatically push calls to voicemail in CME as SRST? One last question : Is Cbarge possible in CME as SRST? I have a hw conf resource registered successfully in SRST mode but it ALWAYS fails to setup the cbarge. Does this look correct? telephony-service sdspfarm units 1 sdspfarm tag 1 BR1CONF no privacy conference hardware srst mode auto-provision none srst ephone template 1 srst ephone description FallBack : Dec 01 2011 16:04:15 srst dn template 1 srst dn line-mode octo em logout 0:0 0:0 0:0 max-ephones 4 max-dn 10 preference 9 ip source-address 10.10.2.1 port 2000 system message Your phone are in fallback mode date-format dd-mm-yy keepalive 10 auxiliary 1 voicemail 2230 max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Dec 01 2011 17:24:55 ephone-dn-template 1 call-forward busy 2230 call-forward noan 2230 timeout 20 huntstop channel 4 ephone-template 1 privacy off privacy-button transfer max-length 4 softkeys remote-in-use CBarge Newcall softkeys idle Redial Newcall Cfwdall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] cisco IOU
Hi, Does CCIE Voice lab routers are real or run on IOU? Guys say not to restart IOU during lab ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cisco IOU
It's real router of course. In actual lab, my router even crashed before Sent from my iPhone Pls pardon my fat fingers. On 2 Dec, 2011, at 11:21 AM, Ken Wyan kew...@gmail.com wrote: Hi, Does CCIE Voice lab routers are real or run on IOU? Guys say not to restart IOU during lab ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.
What are you looking at in the debugs and which leg is this ?
you said you have CUBE in between so you will see 2 seprated H245
negotiation for each leg ,
can you post the H225 and h245 debugs for me please ?
just to make sure that we both talking about the same thing , this
call is Slow start and you have CUBE with Transcoder in it and the
issue you trying to trace is that once you connected the call it got
dropped by the remote GK ?
On the CUBE you have inbound dial-peer with codec G711 and outbound
dial-peer with G729 and then you have transcoder to fix this in the
cube , but on the remote GK you have dial-peer with G711u call only,
if any of the above is not what you have please correct me ,
Ash
On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat ccielab...@gmail.com wrote:
ok, I'm getting to understand this better.
I don't see any mention of a tcs failure though
See the output of debug h245 asn1 below. Where is the indication of a
failure?
Also, I have CUBE running with a Hw transcoder registered locally on HQ
telephony-service.
I would think the CUBE should allocate the xcoder to get around the codec
mismatch.
Output from Debug of H245 ASN1 on HQ/GK/CUBE
Dec 1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0, bytesLeftToDecode =
97
Dec 1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
Dec 1 20:45:36.806:
Dec 1 20:45:36.806: H245 MSC INCOMING PDU ::=
value MultimediaSystemControlMessage ::= request : terminalCapabilitySet :
{
sequenceNumber 1
protocolIdentifier { 0 0 8 245 0 10 }
multiplexCapability h2250Capability :
{
maximumAudioDelayJitter 60
receiveMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FA
HQ#LSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
transmitMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FALSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
receiveAndTransmitMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FALSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
mcCapability
{
centralizedConferenceMC FALSE
decentralizedConferenceMC FALSE
}
rtcpVideoControlCapability FALSE
mediaPacketizationCapability
{
h261aVideoPacketization FALSE
}
logicalChannelSwitchingCapability FALSE
t120DynamicPortCapability FALSE
}
capabilityTable
{
{
capabilityTableEntryNumber 1
capability receiveAudioCapability : g729wAnnexB : 6
},
{
capabilityTableEntryNumber 2
capability receiveAudioCapability : g729AnnexAwAnnexB : 6
},
{
capabilityTableEntryNumber 3
capability receiveAudioCapability : g729 : 6
},
{
capabilityTableEntryNumber 4
capability receiveAudioCapability : g729AnnexA : 6
},
{
capabilityTableEntryNumber 5
capability receiveAndTransmitUserInputCapability : dtmf : NULL
},
{
capabilityTableEntryNumber 6
capability receiveAndTransmitUserInputCapability : basicString :
NULL
},
{
capabilityTableEntryNumber 44
capability receiveAndTransmitUserInputCapability : hookflash :
NULL
}
}
capabilityDescriptors
{
{
capabilityDescriptorNumber 0
simultaneousCapabilities
{
{
1,
2,
3,
4
},
{
5,
6
},
{
44
}
}
}
}
}
Dec 1
