[OSL | CCIE_Voice] Strange CUCM Problem
Hello, So I have the subscriber as primary CPE in the CMgroup and this group, being the only one, is assigned to all devices. I have configured AAR between HQ and branch2. CAC is configured using rsvp. when I dial from HQ to branch2 it says Not enough bandwidth - reorder tone. All other calls to pSTN work ok. You might say that AAR is not enabled but everything is configured corectly. If I stop Call Manager Service on SUB then all devices fall back to PUB and AAR call works. Network congestion rerouting and then BR2 phone rings via PSTN. In other words if SUB is primary CPE - AAR is not invoked and if PUB is primary CPE - AAR works ok. I can say that there is no DB replication problem: Server Number of Replicates Created Replicate_State 192.168.100.168 412 2 - good 192.168.100.169 412 2 - good Any ideas? I just think it's a stupid CUCM on VMWare installation kind of problem. Thanks! ___ 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] CME Option 150
Its advisable you use a loopback interface. Regards Symon On Thu, Aug 23, 2012 at 7:38 PM, Randall Crumm rrcr...@yahoo.com wrote: Hello, If not specified, which interface is it better to configure as the option 150 IP address in the DHCP pool for CME phones? Thanks, Randall ___ 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] calling name support voicemail in srst call-man-fallback
hi folks, for a srst site with call-manager-fallback supports calling name, and when i call the voicemail from the srst phone it doesn't play any voicemail instead it plays CUC general message, is this expected behavior with call-manager-fallback pstn callers are able to leave a voicemail to this phone while in srst... any advice on this matter is appreciated.. thank you krishna.___ 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] calling name support voicemail in srst call-man-fallback
That is not expected behavior. You can achieve integration in SRST versus open trees. Research re-directing ie, voicemail 1XX might be needed under call-manager-fallback in srst. Run Remote Port Status Monitor, etc. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna Sent: Friday, August 24, 2012 9:02 AM To: Online Study Subject: Re: [OSL | CCIE_Voice] calling name support voicemail in srst call-man-fallback hi folks, for a srst site with call-manager-fallback supports calling name, and when i call the voicemail from the srst phone it doesn't play any voicemail instead it plays CUC general message, is this expected behavior with call-manager-fallback pstn callers are able to leave a voicemail to this phone while in srst... any advice on this matter is appreciated.. thank you krishna. itevomcid___ 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] calling name support voicemail in srst call-man-fallback
bill, the point is pstn callers are able to leave voicemail when the site is in SRST, the real problem is retrieving the message from the CUC... i don't understand what does it has to do with alternate extension.. thank you krishna. From: Bill Lake whl...@gmail.com To: Krishna vinayak_...@yahoo.com Sent: Friday, August 24, 2012 9:14 AM Subject: Re: [OSL | CCIE_Voice] calling name support voicemail in srst call-man-fallback did you set up an alternate extension in CUC? On Fri, Aug 24, 2012 at 8:02 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, for a srst site with call-manager-fallback supports calling name, and when i call the voicemail from the srst phone it doesn't play any voicemail instead it plays CUC general message, is this expected behavior with call-manager-fallback pstn callers are able to leave a voicemail to this phone while in srst... any advice on this matter is appreciated.. thank youkrishna. ___ 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] Strange CUCM Problem
Hello Octavian, I had a similar case where using SUB as first option on the CMG for the phones would cause unexpected behaviour... Database appeared OK but something was going on. Sometimes phones couldn't even call to the PSTN. I performed: utils dbreplication repair (sub and pub) utils dbreplication dropadmindb (sub and pub) utils dbreplication reset all And it worked fine again. I also saw the same behaviour with a real deployment in my company, using PUB and SUB on a UCS. We applied the same procedure and issue was solved. I was wondering if it could be a bug. CUCM version in both cases was 8.6. Regards. Daniel Gomez 2012/8/24 Octavian Dumitrescu octavian.dumitre...@datanets.ro Hello, So I have the subscriber as primary CPE in the CMgroup and this group, being the only one, is assigned to all devices. I have configured AAR between HQ and branch2. CAC is configured using rsvp. when I dial from HQ to branch2 it says Not enough bandwidth - reorder tone. All other calls to pSTN work ok. You might say that AAR is not enabled but everything is configured corectly. If I stop Call Manager Service on SUB then all devices fall back to PUB and AAR call works. Network congestion rerouting and then BR2 phone rings via PSTN. In other words if SUB is primary CPE - AAR is not invoked and if PUB is primary CPE - AAR works ok. I can say that there is no DB replication problem: *Server* *Number of Replicates Created* *Replicate_State* 192.168.100.168 412 2 - good 192.168.100.169 412 2 - good ** ** Any ideas? I just think it's a stupid CUCM on VMWare installation kind of problem.*** * Thanks! ** ** __ Information from ESET NOD32 Antivirus, version of virus signature database 7399 (20120819) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.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 -- *Daniel Gómez Quijada* *danie...@gmail.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] H.323 Troubleshooting
Hello guys, I was practicing some troubleshooting today, making some changes in the PSTN router and analysing the SDI traces. And I found out in a codec mismatch situation, the trace is different if I configure the trunk as an H.323 gateway or as an Inter Cluster Trunk (non GK controlled). If I configure an ICT, I can clearly see the incoming TCS from the PSTN supporting g711u, then the outgoing TCS from CUCM saying it supports g729 only. And finally the PSTN sends a terminalCapabilitySetReject, and the call is released with cause code C1 (which is Bearer capability not implemented). But if I configure an H.323 gateway, I can only see the incoming TCS from the PSTN saying it supports g711u only, and after that, the call is released. CUCM does not send its TCS, and I cannot see any terminalCapabilitySetReject message. The cause code now is AF (which is Resources unavailable, unspecified: The channel or service that the user requests is unavailable for an unknown reason. This problem is usually temporary.) Is this the correct behavior? So how can I explain a codec mismatch situation in a H.323 gateway, since no message clearly says that? Thanks, Bruno ___ 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] H.323 Troubleshooting
Please correct me if necessary. I've done some limited home testing on ICT vs Device Gateway. My understanding from the SRND is that Device Trunk ICT Non-gatekeeper should only be used with another CallManager/Cluster. With Trunk H323 Non-Gatekeeper Controlled TCS is always sent. I'm also wondering if anything non-standard is sent as it expects the other end to be a CallManager/Cluster. With Device Gateway the default is unchecked Wait for Far End H.245 Terminal Capability Set. Depending on who becomes Master/Slave you might not see a TCS Reject but instead the Master (be sure to understand MSD and be able to tell who is Master) sending a disconnect without a definitive proof of why. I would then check Wait for Far End H.245 Terminal Capability Set and review SDI/SDL traces for TCS problems, however if the other side doesn't send TCS I have a Stalemate and can't show proof TCS is the problem, only how I did my troubleshooting that problem is far end configuration. References http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/callpros.html ICT Non-Gatekeeper Controlled http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html#wp1044701 http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg27089.html http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/video.html#wp1046354 Note: For intercluster trunks and gatekeeper-controlled intercluster trunks, the Wait for Far-End to Send TCS option is always disabled and cannot be enabled. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bruno Nonogaki Sent: Friday, August 24, 2012 2:11 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] H.323 Troubleshooting Hello guys, I was practicing some troubleshooting today, making some changes in the PSTN router and analysing the SDI traces. And I found out in a codec mismatch situation, the trace is different if I configure the trunk as an H.323 gateway or as an Inter Cluster Trunk (non GK controlled). If I configure an ICT, I can clearly see the incoming TCS from the PSTN supporting g711u, then the outgoing TCS from CUCM saying it supports g729 only. And finally the PSTN sends a terminalCapabilitySetReject, and the call is released with cause code C1 (which is Bearer capability not implemented). But if I configure an H.323 gateway, I can only see the incoming TCS from the PSTN saying it supports g711u only, and after that, the call is released. CUCM does not send its TCS, and I cannot see any terminalCapabilitySetReject message. The cause code now is AF (which is Resources unavailable, unspecified: The channel or service that the user requests is unavailable for an unknown reason. This problem is usually temporary.) Is this the correct behavior? So how can I explain a codec mismatch situation in a H.323 gateway, since no message clearly says that? Thanks, Bruno itevomcid___ 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
