[OSL | CCIE_Voice] cBarge during SRST
Hi All, Let say we need to make sure the same call feature remains during SRST, and there is a share line with cbarge during normal CUCM operation. However, if the requirement is Do not pre-define any ephone or ephone-dn in running-config how would you interupt this? The simplest way is to use srst mode auto-provision none. The issue with this command is that cBarge during SRST would not work. So if we use srst mode auto-provision all, the configure in running-config will be learnt during SRST. would that still consider as pre-define? Cheers ___ 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] LAN QoS
Hi all Just want to get some advise regarding LAN QoS. Let say if the question ask to apply priority queue and specify certain drop % based on a COS value to the interface that connect to the router. What would you do with the interface that connecting to the phone? The reason I asked is because if auto-qos is used on the phone ports, a few config line will be generated, such as policy map etc. Do you normally leave those in there on the phone ports? Cheers Vega ___ 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] Lab Grading
Hi all First of all, let me clarify that I am not trying to break NDA or trying to sell anything here, just something I cant get my head around with. I just had my second attmpt yesterday, and I failed. What really demoralising for me is that I thought I was fully prepared. What even more demoralising is that when I looked at the score card, I didnt get 100% in the area I thought I would get a 100%. I dont mean to say I know it all (especially now), but there are areas on my score card that really shocks me. As I read through the exam, I know what the questions were requiring (at least I thought I did). For example, we all know that the topics of DHCP, NTP, VLAN will be in the network infrastructure area. And what could go wrong in those areas? I mean, if you dont setup those area correctly, all your subsequence config will has problem right? But yet, I dont get full marks in those. Similarly the gateway area, QoS area etc. Right now I am pretty lost as to what I can prepare or study on, or at least how to check my config? I know this sounds really bad (or arrogent) but I really have no idea as how to confirm the config I did can get the full mark in those area. I guess I am hoping if someone that share their strategy as to how to confirm their work is good or fulfilled the requirement? hope that makes sense? ___ 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] Vol2 Lab8 Question5.6
Hi Baktha The questions said to only police the RTP traffic that were remarked by RSVP CAC. So when calls that is allowed by RSVP, the RTP is still marked as EF. When calls that fail by RSVP CAC, the RTP will be remarked as cs1. It means that there are cases when RTP is EF (as normal) or CS1 (remarked by RSVP CAC), I understand when using AutoQos, it automatically generate a class-map as such: class-map match-any RTP match ip protocol rtp auto match access-group RTP This class-map should receive the priority treatment as per normal. now if I add another class map as follow class-map match-all remarkRTP match ip dscp cs1 So the policy-map looks something like this policy-map LLQ class RTP priority 18 class remarkRTP police rate percent 33 now my question is, for those RTP that were remarked as CS1, which class will it goes under?, it seems to fit both classes. --- On Sun, 5/2/12, Baktha Muralidharan muralic...@gmail.com wrote: From: Baktha Muralidharan muralic...@gmail.com Subject: Re: [OSL | CCIE_Voice] Vol2 Lab8 Question5.6 To: ccie_voice@onlinestudylist.com Received: Sunday, 5 February, 2012, 2:55 PM can't you use a class-map with match-any as follows? class-map match-any foo match access-group rtpports match ip dscp cs1 thanks, /Baktha Message: 3 Date: Sat, 4 Feb 2012 00:18:28 -0800 (PST) From: Vega Wong vega2...@yahoo.com.au To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol2 Lab8 Question5.6 Message-ID: 1328343508.576.yahoomailclas...@web65902.mail.ac4.yahoo.com Content-Type: text/plain; charset=utf-8 Hi experts I am working on question 5.6 in Vol 2 Lab 8. But I dont understand how the DSG explains the answers. Questions: HQ and BR1 - 768kbps PQ for 4 calls 5% for signalling mark the excess RTP traffic (failed RSVP) to 33%. (this type of RTP been marked as CS1 in earlier question) Ensure FRF.12 LFI is being used. The DSG is saying we should not use auto qos voip trust for branch 1, due to the limitation on HWIC-ESW. However, if I do that, the class-maps created will be matching the traffic based on the port. Here is what I am confused, one of the class-map created is matching the RTP traffic by the ports. Then if I need to catch the RTP that was marked as CS1 by CUCM, I will need to create another class-map right? but how does the policy-map know which class it should put the RTP traffic in? In another words, some of the RTP traffic will be matched by two different classes. Which class would take precedences? On the other hand, if we are not to trust the phone connecting to the HWIC, should we be creating a service policy and apply it as a service-policy input on the HWIC port to mark all the traffic coming in to the branch router? Can anyone shed some light on this one? Cheers Vega -Inline Attachment Follows- ___ 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] Vol2 Lab8 Question5.6
Hi experts I am working on question 5.6 in Vol 2 Lab 8. But I dont understand how the DSG explains the answers. Questions: HQ and BR1 - 768kbps PQ for 4 calls 5% for signalling mark the excess RTP traffic (failed RSVP) to 33%. (this type of RTP been marked as CS1 in earlier question) Ensure FRF.12 LFI is being used. The DSG is saying we should not use auto qos voip trust for branch 1, due to the limitation on HWIC-ESW. However, if I do that, the class-maps created will be matching the traffic based on the port. Here is what I am confused, one of the class-map created is matching the RTP traffic by the ports. Then if I need to catch the RTP that was marked as CS1 by CUCM, I will need to create another class-map right? but how does the policy-map know which class it should put the RTP traffic in? In another words, some of the RTP traffic will be matched by two different classes. Which class would take precedences? On the other hand, if we are not to trust the phone connecting to the HWIC, should we be creating a service policy and apply it as a service-policy input on the HWIC port to mark all the traffic coming in to the branch router? Can anyone shed some light on this one? Cheers Vega ___ 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] IPX vol1 Lab9a
Hi I found it easier to understand or familise with the configuration steps if practise IPMA and Extension Mobilty separately. As for the URL, you can find them in the CUCM SRND. Cheers --- On Wed, 25/1/12, Jeferson Guardia jefers...@gmail.com wrote: From: Jeferson Guardia jefers...@gmail.com Subject: Re: [OSL | CCIE_Voice] IPX vol1 Lab9a To: Boris K boris.k...@gmail.com Cc: OSL Voice ccie_voice@onlinestudylist.com Received: Wednesday, 25 January, 2012, 2:07 PM Just found it. Good deal!! :-) Cisco Unified IP Phone Service Configuration Add the Cisco IP Manager Assistant service as a new Cisco Unified IP Phone Service. Configure a name, description, and the URL for the Cisco IP Manager Assistant service. The name and description that you enter should be in the local language because it displays on the manager Cisco Unified IP Phone. For more information, see IP Phone Services Configuration in the Cisco Unified Communications Manager Administration Guide. Provide a URL by using the format http://server-ipaddress:8080/ma/servlet/MAService?cmd=doPhoneServiceName=#DEVICENAME# For example http://123.45.67.89:8080/ma/servlet/MAService?cmd=doPhoneServiceName=#DEVICENAME# Configuration Tips To provide redundancy for the Cisco Unified IP Phone Service, create a Cisco Unified IP Phone Service that uses the host name rather than the IP address. The host name in DNS should resolve to both Cisco Unified Communications Manager Assistant primary and backup IP addresses. The phone functionality for softkeys and filtering, as well as the phone service, will fail over automatically in the case of a failover. 2012/1/25 Boris K boris.k...@gmail.com hi mate, you can get IPMA and EM URLs from the CUCM Help page, just search for DEVICENAME UCCX Agent Login and IPPM URLs you need to memorise or remember shortcuts on Cisco site. Regards, Boris On Wed, Jan 25, 2012 at 1:34 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: No brother... The best way is to find and learn where it is on Cisco DocCD... :) Emanuel Damasceno CCNP Voice On Wed, Jan 25, 2012 at 12:09 AM, Jeferson Guardia jefers...@gmail.com wrote: IPMA complex, too many steps.. but somehow I feel after practicing for a while I will get used to it. IPX covers well but for some steps, even watching the video, it doesnt take you where you need a piece of specific information to accomplish a task, example: Adding IP Phone Services for IPMA and EM (Extension Mobility) , I know how to create but the URL string I should create, it doesnt mention where to find it or the full string for it. After googling it, I found the URL. Anyone would have a better way to find these URL`s in case in the future I have to deal with adding new IP Phone Services? This was new to me. Thank you all!! -- Jeferson Guardia CCIE #28157 ___ 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 -- Jeferson Guardia CCIE #28157 -Inline Attachment Follows- ___ 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] Voicemail access during AAR
Hi experts I am working on Vol 2. Lab 7 and try to fully understand the topic of voice mail access during AAR, but I am struggling to have a clear picture. Here is the setup HQ GW - MGCP BR GW - H323 HQ Phones, and BR Phones are both SCCP. The line on both phone been assigned to the same AAR group. Only HQ phone assigned with AAR group and CSS AAR on the device level. voicemail for BR is on CUE which is integrated with CUCM using CTI route point. All integration is configured and tested ok AAR between the phones are working, the HQ Phone will reroute out to PSTN to reach BR Phones when there is not enough bandwidth. Also, HQ Phone can directly dial in to the voicemail pilot (reroute out to PSTN) and reach the CUE log in. BR phone can press the message button and reach the sign-in prompt for CUE (only asking for PIN) However, when HQ phone calls BR phone and BR Phone doesnt pick up, just as it should transfer to voicemail, I get fast busy tone on HQ Phone. I am trying to understand, at this instance, I am still using the AAR CSS on HQ Phone to reach the voice mail pilot right? I imagine if HQ Phone can successfully call to voicemail directly during AAR, it shouldnt be different when it is transferred by BR Phone? Also, I am trying to understand when do we need to assign AAR group and AAR CSS to the gateway? and why? Please help Thanks in advance Vega ___ 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] SRST keepalive time
Hi there Actually, as this is a real live customer, so perhaps suggest to the customer that he can run his remote sites as CME, and setup a trunk between the CUCM at the main site and the CMEs at the remote sites. Of course there will be implication in regards to the dial plan and other configuration, but since he is on a crappy link that causing the phone system instable, may as well change the architecture to fit the scenario. --- On Wed, 9/11/11, Emanuel Damasceno aedamasc...@gmail.com wrote: From: Emanuel Damasceno aedamasc...@gmail.com Subject: Re: [OSL | CCIE_Voice] SRST keepalive time To: ccie_voice@onlinestudylist.com Received: Wednesday, 9 November, 2011, 12:18 PM Thanks for the replies. It didn't work. The customer has a centralized solution with 2 sites. His two sites use MGCP, but unfortunately his environment is a screwy solution. He has EVERYTHING on VLAN 1. I explained to him many times he needed to segregate his VLANs and use QoS for his WAN. But we have to work with this CRAPPY solution for the time being... This is what he needs (I'll try to be as specific as I can). His screwed up WAN link is a 2MBps (I highly doubt this) without any QoS applied (neither in his side or his SP). The link is very unstable and his phones on the remote site keeps coming back and forth from SRST. Since it keeps doing it for so long, the phones keep just being out of its service, because it registers, then there is a problem with the link and all his phones in the remote site tries to go to SRST mode, but then all of the sudden the link is back and the phones try to go back to CUCM. He was asking me to put a 12 hour (yes, who doesn't have a customer like this...) before the phones tried to connect with CUCM again. In other words, he wanted to let the phones in SRST mode for a few hours until he fixes his problem with the link. So, I tried the command ccm-manager switchback uptime-delay minutes, and we simulate a WAN outage. The phones entered SRST (well, at least he said it was. I am the remote support), but when he turned on the CUCM back, the phones registered right away. Even after setting the time for two hours. So, we went on Enterprise Parameters and changed the connection duration monitor but I didn't do it clusterwise. I did it on the device pool on the remote site. Now, the phones didn't register to SRST so none of the phones were making any calls or receiving it (that's why I wonder his SRST solution never worked in the first place). So I rolled back to the standard settings and I told him I'd study his solution and give him back an answer. Honestly, knowing what he has, I think this won't work, but can you guys give me any ideas? I appreciate it. Antonio Emanuel Damasceno CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written) CompTIA Network+ On Tue, Nov 8, 2011 at 2:44 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: Hello experts, I currently have a customer who has CUCMBE in his environment with SRST enabled. His voice gateway is over a WAN link, and the link is unstable. His phones keep registering back and forth, and now he wants to keep his phones in SRST mode for a little longer than usual. His configs are call-manager-fallback based (no CME as SRST)... How can I achieve this? Is it through CUCMBE or his CME? His gateway is MGCP. Thanks Antonio Emanuel Damasceno CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written) CompTIA Network+ -Inline Attachment Follows- ___ 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] LAN QoS egress queue setting
Hi experts I am working on the topic of LAN QoS. I understand we will be given tasks to modify the setting on the egress queues of the switches. The setting includes, buffer sizes, shape and share setting etc. My question is, is there a best practise on what values these figures should change to? for example, if the task ask to change the buffer size to 50% for queue 2, what should be the buffer size set to for Q1, Q3 and Q4 in the queue-set? 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
[OSL | CCIE_Voice] H323 debug messages
Hi experts I am trying to practise different scenario with H323 gateways and gatekeepers. Does anyone have any good reference to read those output from debug h225 asn1 ?? 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] H323 debug messages
Hi I am trying to work out the meaning of these “terminationcause” from the DRQ. They are in the form of 08028081 or 08028091, is there anywhere I can find out the meaning of these code? Thanks From: Amit Singh [mailto:batraji...@yahoo.com] Sent: Tuesday, October 04, 2011 6:32 PM To: vega2...@yahoo.com.au Subject: Re: [OSL | CCIE_Voice] H323 debug messages iOS 12.4 troubleshooting guide. Sent from my iPad On 4/10/2011, at 6:52 PM, Vega Wong v...@iinet.net.au wrote: Hi experts I am trying to practise different scenario with H323 gateways and gatekeepers. Does anyone have any good reference to read those output from debug h225 asn1 ?? 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 ___ 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] MOH with H323 gateway
Hi Experts I finally got this to work, after I have removed all the multicast routing command in HQ GW. I can hear two separate moh streams on the PSTN phone. one from the router flash and the other one from CUCM. So i used an ACL to block out the stream from CUCM, I am using the ACL to filter out UPD port 16384 to 32676 from the two CUCMs. As expected, I can only hear the single moh stream from the router flash. So just one question, if I filter out all the RTP traffic from CUCM, will I run into trouble? Such as blocking annunicator or other thing? Please advice On Sun, 2/10/11, Jason Langenfeld jlangenf...@prosysis.com wrote: From: Jason Langenfeld jlangenf...@prosysis.com Subject: Re: [OSL | CCIE_Voice] MOH with H323 gateway To: Bill Lake whl...@gmail.com, vega2...@yahoo.com.au vega2...@yahoo.com.au Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Received: Sunday, 2 October, 2011, 3:53 AM Check the MRGL setting on HQ Device Pool is correct, then if it is, try manually adding the MRGL on the h323 gateway in UCM. From: Bill Lake whl...@gmail.commailto:whl...@gmail.com Date: Sat, 1 Oct 2011 07:31:15 -0500 To: vega2...@yahoo.com.aumailto:vega2...@yahoo.com.au Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH with H323 gateway Try an ACL maybe something like this would work HQ-3560(config)#ip access-list extended 100 HQ-3560(config-ext-nacl)#deny ip host 239.1.1.1 any HQ-3560(config-ext-nacl)#permit ip any any HQ-3560(config-ext-nacl)#exit HQ-3560(config)#interface fastethernet x/x/x HQ-3560(config-if)#ip access-group 100 in apply it to the up link interface, now I don't do ACL's very often so I could easily have this wrong, you might have to configure it on the router and have it block on the incoming port from the switch. This should stop all ip 239.1.1.1 traffic. On Sat, Oct 1, 2011 at 2:16 AM, Vega Wong vega2...@yahoo.com.aumailto:vega2...@yahoo.com.au wrote: Hi I have investigated further into this issue, and I think I may have found the reason for this. As the MOH server is setup with 1 hop limit with the multicast setting, and the HQ router/GW is only 1 hop away, so the moh stream from CUCM can reach to the loopback address of the HQ router within 1 hop. But correct me if I am wrong, that could be the reason why the moh stream from CUCM can reach PSTN caller. So my question is, is there a way to make the HQ loopback more than 1 hop away? Thanks From: Bill Lake [mailto:whl...@gmail.commailto:whl...@gmail.com] Sent: Friday, September 30, 2011 3:06 AM To: Vega Wong Subject: Re: [OSL | CCIE_Voice] MOH with H323 gateway Hey maybe you were always getting unicast. so you might want to check that on the H323 gateway. You always do the same IP as the CUCM is, so for CUCM set as ip it would be 239.1.1.1 by default for G711 on the first CUCM set as MOH, the second one would have to be set to a different IP like 239.2.1.1 or 239.1.2.1 as CUCM increments for each codec on the last octet. SO your CUCM is working to Multicast so you must have your music file on it checked as multicast, the PUB MOH server checked and your MRG/L set to use it. Otherwise the phones at BR1 would not play the spoofed Multicast music and your HQ would not have any devices spoofed to use multicast. Also if you CUCM is telling a device to use the multicast it should not also be using Unicast so maybe this is a bug or something I just don't know. Maybe there is a way to shut down the unicast stream, just google it and see but otherwise ask again the entire list and see if maybe someone has seen this before. On Thu, Sep 29, 2011 at 8:48 AM, Vega Wong vega2...@yahoo.com.aumailto:vega2...@yahoo.com.au wrote: Yes, this is very very confusing (and frastrasting for me). But you actually brought up a very good point. Looks like the PSTN caller is actually getting unicast moh from CUCM. My reason is that when the call get through, I get this: HQ-RTR#show call active voice | inc Remote RemoteIPAddress=10.10.210.11 RemoteUDPPort=24578 RemoteSignallingIPAddress=10.10.210.11 RemoteSignallingPort=33131 RemoteMediaIPAddress=10.10.210.10 RemoteMediaPort=24578 10.10.210.10 is the Pub, I think I am getting unicast moh from CUCM, this probably the reason why I am getting the two streams mixed when I turned the router into a mgcp gateway. The output of the above command at Br1 shows that the Remote Medial address as 239.1.1.1. For this kind of setup, should the multicast moh command at both gateway using the same multicast IP 239.1.1.1 ? Correct me if I am wrong, should one of them be using 239.2.1.1? mind that there is only one MOH server. Also, should I do anything on the switches? for example, turn on or off ip igmp snooping? Thanks again for your suggestion --- On Thu, 29/9/11, Bill Lake whl...@gmail.commailto:whl...@gmail.com wrote: From: Bill Lake whl...@gmail.commailto:whl...@gmail.com Subject
Re: [OSL | CCIE_Voice] MOH with H323 gateway
call and put the call on hold: HQ-RTR#show call active voice | inc Remote RemoteIPAddress=10.10.210.11 RemoteUDPPort=24578 RemoteSignallingIPAddress=10.10.210.11 RemoteSignallingPort=33131 RemoteMediaIPAddress=10.10.210.10 RemoteMediaPort=24578 I can hear moh on the PSTN phone, but the moh stream is actually coming from the CUCM (the default music files) ! This is when I make a PSTN call via the mgcp gw and put the call on hold: BR1-RTR#show call active voice | inc Remote RemoteIPAddress=0.0.0.0 RemoteUDPPort=16384 RemoteSignallingIPAddress=0.0.0.0 RemoteSignallingPort=0 RemoteMediaIPAddress=239.1.1.1 RemoteMediaPort=16384 With the mgcp gw, I can hear the moh stream from router (my own music file). I am really confused here, I just cant work out why I cant get the h323 to stream the moh to the PSTN caller as the mgcp gw does. Thanks guys --- On Wed, 28/9/11, Vega Wong vega2...@yahoo.com.au wrote: From: Vega Wong vega2...@yahoo.com.au Subject: Re: [OSL | CCIE_Voice] MOH with H323 gateway To: ccie_voice@onlinestudylist.com, DeShon Crayton dcrayto...@comcast.net, whl...@gmail.com Received: Wednesday, 28 September, 2011, 8:50 AM Hi guys Thanks for the suggestion. further info with the setup: The H323 GW is the HQ router, local to the CUCM. I have already configured the command in the H323 GW: ip multicast-routing (global) ip pim dense-mode (voice interface and loopback) The correct file is setup with the moh command under telephony-service, as well as the multicast moh 239.1.1.1 port 16384 route 10.10.200.3 10.10.110.1 (the two address are the IP for the voice interface and loopback) What really puzzles me is that when the two IP Phones (both HQ) call each other, if one of them is put on hold, the moh is stream from the router (my own music file). I have also checked the phone URL, it shows the correct multicast IP (239.1.1.1). So I suppose that prove my multicast moh setting on the router is correct? The problem is when I call PSTN from one of these two HQ phones, (or PSTN calls to these phones). The PSTN phone will get the moh stream from CUCM rather than from the HQ router. I am at work at the moment, but I can send through the config for the HQ router later today Thanks again --- On Wed, 28/9/11, DeShon Crayton dcrayto...@comcast.net wrote: From: DeShon Crayton dcrayto...@comcast.net Subject: RE: [OSL | CCIE_Voice] MOH with H323 gateway To: 'Vega Wong' vega2...@yahoo.com.au, ccie_voice@onlinestudylist.com Received: Wednesday, 28 September, 2011, 8:20 AM Hello Vega, There could be many reasons as to why the MOH for the h323 gw is not working properly. Can you forward your config.. I would check the following. 1. Make sure the h323 gw has ccm-manager music-on-hold2. Make sure that the moh file name is correctly named with telephony-service or call-manager-fallback3. Try command no ip igmp snooping4. Make the multicast is enabled on the appropriate router and appropriate interfaces5. Reboot the router From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vega Wong Sent: Tuesday, September 27, 2011 9:24 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MOH with H323 gateway Hi Experts I am doing some testing with streaming multicast MOH from the router flash, but I encounter a strange issue. Hoping someone can point me to the right direction to tackle this. I have setup two sites - HQ and BR1. HQ is setup with H323 gateway, BR1 is setup with mgcp gateway. I have setup two device pools, they both using the same MGRL and same region, (the reason for the two device pools is because I want them to display different timezone). All phones are registered to CUCM - only one MOH server (Pub) is assigned to MRG and multicast enabled. The MOH server (Sub) is not registered - I changed the Run Flag to false on that MOH server. For both site, I can get the multicast moh stream from the router when an IP phone call another IP Phone. I have put my own moh file in the router, so straightaway I can tell its from the router. I have also confirm on the IP Phone URL as well as show call active voice | be Remote that the remote IP is 239.1.1.1 With the mgcp gateway, when I put the PSTN call on hold, the PSTN phone can hear the moh from the router. (again confirmed with show call active voice | be Remote). But for H323, when the PSTN call is put on hold, it get the moh stream from the CUCM server! The remoteIP is the IP of the CUCM (Unicast?) Can someone explain to me the reason for PSTN call through mgcp can turn to the multicast moh, but not through H323 gateway? Thanks in advance -Inline Attachment Follows- ___ 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
Re: [OSL | CCIE_Voice] MOH with H323 gateway
Thanks everyone for the suggestion. I am really keen to get to the bottom of this, so please tell me what other information I should provide? further to earlier question, the moh file is definitely the right format, because I am using the exact same file in the branch (mgcp) gateway, and it is working perfectly fine. What really throws me off is, within this problemtic HQ site, when two IP phones call each other and put on hold, I can hear the correct moh stream from the router. So I suppose that proves the setting of multicast and 239.1.1.1 is correct. the only issue is with PSTN callers, the PSTN phone is getting the moh stream from CUCM. --- On Thu, 29/9/11, Bill Lake whl...@gmail.com wrote: From: Bill Lake whl...@gmail.com Subject: Re: [OSL | CCIE_Voice] MOH with H323 gateway To: Ashraf Ayyash ash.ayy...@gmail.com Cc: DeShon Crayton dcrayto...@comcast.net, Vega Wong vega2...@yahoo.com.au, ccie_voice@onlinestudylist.com Received: Thursday, 29 September, 2011, 8:36 AM I would agree that we could use more information but if he removed ip pim dense-mode, wouldn't he stop the CUCM from flooding, this would stop the MOH from CUCM from being available to be used and he would either not have any MOH or would have his MOH from the flash on the router. On Wed, Sep 28, 2011 at 4:16 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello All , in the explination you gave about the problem you must know what type of moh you got on the HQ and you can confirm that by hear the moh and you will know if its MMOH or unicast and this is very important to figure out because this will turn the troubleshooting to diff direction , also if you have connected call betweeen HQ and BR1 Phone and BR1 phone place HQ on hold what you will get ? in regard of the Multicast config changes , i dont see any reason for that , Note that he is sourcing MOH from the FLASH which mean we are not doing any MMOH routing here , we are flooding the MOH to the specified route under the telephone or call-manager fallback so in this case we dont need any kind of Multicasting setup from the provided info i cannot go anywhere in the troubelshoting , you have to send more detailed info about this issue Thanks Ash On Wed, Sep 28, 2011 at 5:53 AM, DeShon Crayton dcrayto...@comcast.net wrote: Also, confirm that the UCM moh server is using multicast address 239.1.1.1 and incrementing on ip address. From: DeShon Crayton [mailto:dcrayto...@comcast.net] Sent: Wednesday, September 28, 2011 8:51 AM To: 'Vega Wong'; 'ccie_voice@onlinestudylist.com'; 'whl...@gmail.com' Subject: RE: [OSL | CCIE_Voice] MOH with H323 gateway Hello Vega, I would add the following: Config tno ip igmp snopping int l0ip pim dense-mode int fa 0/20ip pim dense-mode Confirm that “MOH_CL.wav” is in flashConfirm that “MOH_CL.wav” is properly formatted to be used by the cisco router. Try using the default “music-on-hold.au” that comes with CME for testing purposes.Reboot the router.. From: Vega Wong [mailto:vega2...@yahoo.com.au] Sent: Wednesday, September 28, 2011 7:29 AM To: ccie_voice@onlinestudylist.com; DeShon Crayton; whl...@gmail.com Subject: Re: [OSL | CCIE_Voice] MOH with H323 gateway Hi guys I have attached more info for this, hope you can help: ! H323 gw config ! hostname HQ-RTR ! network-clock-participate slot 1 ! dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ip multicast-routing no ipv6 cef ! multilink bundle-name authenticated ! ! isdn switch-type primary-ni ! ! ! voice-card 0 ! voice-card 1 ! ! ! controller T1 1/0/0 pri-group timeslots 1-3,24 ! controller T1 1/0/1 ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ip pim sparse-dense-mode ! interface GigabitEthernet0/0 no ip address duplex auto speed auto ! interface GigabitEthernet0/0.10 encapsulation dot1Q 10 ip address 10.10.100.1 255.255.255.0 ! interface GigabitEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.11 ip pim sparse-dense-mode h323-gateway voip bind srcaddr 10.10.200.3 ! interface GigabitEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/2/0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/2/0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/2/0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! interface Serial0/2/1 no ip address shutdown clock rate 200 ! interface Serial1/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice no cdp enable ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0
[OSL | CCIE_Voice] MOH with H323 gateway
Hi Experts I am doing some testing with streaming multicast MOH from the router flash, but I encounter a strange issue. Hoping someone can point me to the right direction to tackle this. I have setup two sites - HQ and BR1. HQ is setup with H323 gateway, BR1 is setup with mgcp gateway. I have setup two device pools, they both using the same MGRL and same region, (the reason for the two device pools is because I want them to display different timezone). All phones are registered to CUCM - only one MOH server (Pub) is assigned to MRG and multicast enabled. The MOH server (Sub) is not registered - I changed the Run Flag to false on that MOH server. For both site, I can get the multicast moh stream from the router when an IP phone call another IP Phone. I have put my own moh file in the router, so straightaway I can tell its from the router. I have also confirm on the IP Phone URL as well as show call active voice | be Remote that the remote IP is 239.1.1.1 With the mgcp gateway, when I put the PSTN call on hold, the PSTN phone can hear the moh from the router. (again confirmed with show call active voice | be Remote). But for H323, when the PSTN call is put on hold, it get the moh stream from the CUCM server! The remoteIP is the IP of the CUCM (Unicast?) Can someone explain to me the reason for PSTN call through mgcp can turn to the multicast moh, but not through H323 gateway? Thanks in advance ___ 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] MOH with H323 gateway
Hi guys Thanks for the suggestion. further info with the setup: The H323 GW is the HQ router, local to the CUCM. I have already configured the command in the H323 GW: ip multicast-routing (global) ip pim dense-mode (voice interface and loopback) The correct file is setup with the moh command under telephony-service, as well as the multicast moh 239.1.1.1 port 16384 route 10.10.200.3 10.10.110.1 (the two address are the IP for the voice interface and loopback) What really puzzles me is that when the two IP Phones (both HQ) call each other, if one of them is put on hold, the moh is stream from the router (my own music file). I have also checked the phone URL, it shows the correct multicast IP (239.1.1.1). So I suppose that prove my multicast moh setting on the router is correct? The problem is when I call PSTN from one of these two HQ phones, (or PSTN calls to these phones). The PSTN phone will get the moh stream from CUCM rather than from the HQ router. I am at work at the moment, but I can send through the config for the HQ router later today Thanks again --- On Wed, 28/9/11, DeShon Crayton dcrayto...@comcast.net wrote: From: DeShon Crayton dcrayto...@comcast.net Subject: RE: [OSL | CCIE_Voice] MOH with H323 gateway To: 'Vega Wong' vega2...@yahoo.com.au, ccie_voice@onlinestudylist.com Received: Wednesday, 28 September, 2011, 8:20 AM Hello Vega, There could be many reasons as to why the MOH for the h323 gw is not working properly. Can you forward your config.. I would check the following. 1. Make sure the h323 gw has ccm-manager music-on-hold2. Make sure that the moh file name is correctly named with telephony-service or call-manager-fallback3. Try command no ip igmp snooping4. Make the multicast is enabled on the appropriate router and appropriate interfaces5. Reboot the router From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vega Wong Sent: Tuesday, September 27, 2011 9:24 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MOH with H323 gateway Hi Experts I am doing some testing with streaming multicast MOH from the router flash, but I encounter a strange issue. Hoping someone can point me to the right direction to tackle this. I have setup two sites - HQ and BR1. HQ is setup with H323 gateway, BR1 is setup with mgcp gateway. I have setup two device pools, they both using the same MGRL and same region, (the reason for the two device pools is because I want them to display different timezone). All phones are registered to CUCM - only one MOH server (Pub) is assigned to MRG and multicast enabled. The MOH server (Sub) is not registered - I changed the Run Flag to false on that MOH server. For both site, I can get the multicast moh stream from the router when an IP phone call another IP Phone. I have put my own moh file in the router, so straightaway I can tell its from the router. I have also confirm on the IP Phone URL as well as show call active voice | be Remote that the remote IP is 239.1.1.1 With the mgcp gateway, when I put the PSTN call on hold, the PSTN phone can hear the moh from the router. (again confirmed with show call active voice | be Remote). But for H323, when the PSTN call is put on hold, it get the moh stream from the CUCM server! The remoteIP is the IP of the CUCM (Unicast?) Can someone explain to me the reason for PSTN call through mgcp can turn to the multicast moh, but not through H323 gateway? Thanks in advance ___ 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] Problem with UC and CUCME integration
Actually I sorted it out, I didnt set the CSS in the CTI Route point and CTI port. I outlooked it. Thanks :) --- On Sat, 10/9/11, Ashraf Ayyash ash.ayy...@gmail.com wrote: From: Ashraf Ayyash ash.ayy...@gmail.com Subject: Re: [OSL | CCIE_Voice] Problem with UC and CUCME integration To: Vega Wong vega2...@yahoo.com.au Cc: Rynard Coetzee rynard.coet...@bytes.co.za, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Received: Saturday, 10 September, 2011, 7:43 PM from where you have choose unsolicited in the CUC ? what you need to have configured in case of SIP integration with CUCM-CUC is : from System Security Profile SIP Trunk Security Profile. check (enable) Accept Unsolicited Notifications so that Cisco Unity and Unity Connection can notify Unified CM of Message Waiting Indicator (MWI) events. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/vmessage.html and make sure that the SIP trunk : Out-Of-Dialog Refer Calling Search Space SUBSCRIBE Calling Search SpaceRerouting Calling Search Space ## can see the phones Partition , if this is done already , can you collect sniffer traces from the CCM and check why the unsolicited have been declined by the CCM (you will see reason code inside the sip response message for the mwi notify ? CCM SDI/SDL traces along with Sniffer traces will give you the whole story , and if you need help with the traces just bring them up here and we can take a look and comment . Ash On Fri, Sep 9, 2011 at 6:07 PM, Vega Wong vega2...@yahoo.com.au wrote: Hi team Can you tell me where I can check the MWI setting in UC whether it is notified or Unsolicited? I tried to configure as unsolicited but it doesnt work. Thanks team --- On Tue, 6/9/11, Ashraf Ayyash ash.ayy...@gmail.com wrote: From: Ashraf Ayyash ash.ayy...@gmail.com Subject: Re: [OSL | CCIE_Voice] Problem with UC and CUCME integration To: Rynard Coetzee rynard.coet...@bytes.co.za Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Received: Tuesday, 6 September, 2011, 12:33 AM glad to hear its working , have a good study/day Ash On Mon, Sep 5, 2011 at 4:47 PM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Ash You were spot on mate ,it was the server on the port group page that was pointing to the incorrect IP ,was pointing to Loopback address where my SIP was bound to previously ,I had changed the binding to the voice vlan but forgot to change it on UC. Working 100% now. Thanks for the help. -Original Message- From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com] Sent: 05 September 2011 03:16 PM To: Rynard Coetzee Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with UC and CUCME integration Hello , the CME is sending 2 invite to the CUC and the CUC havnt accept the call with routing failed issue : Warning: 399 'Routing failed: socket=10.10.210.12:5060' review your config once again , why the CME sending 2 invite with 2 diff ips ? and make sure that the server you added from the port group page in the CUC have the same ip which is the IP of your Vlan400 . Ash On Mon, Sep 5, 2011 at 2:56 PM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: *Sep 5 12:50:41.175: //-1//SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:3600@10.10.202.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.202.51:5060;branch=z9hG4bK14031c57 From: BR2Ph1 sip:3001@10.10.202.1;tag=fcfbfbcb61d6002086bfb221-0332f79c To: sip:3600@10.10.202.1 Call-ID: fcfbfbcb-61d60013-326b97fb-35906d9e@10.10.202.51 Max-Forwards: 70 Date: Mon, 05 Sep 2011 12:50:41 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7961G/8.5.3 Contact: sip:3001@10.10.202.51:5060;transport=udp Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: BR2Ph1 sip:3001@10.10.202.1;party=calling;id-type=subscriber;privacy=off;sc reen=yes Allow-Events: kpml,dialog Content-Length: 276 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 14250 0 IN IP4 10.10.202.51 s=SIP Call t=0 0 m=audio 18866 RTP/AVP 0 8 18 101 c=IN IP4 10.10.202.51 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv *Sep 5 12:50:41.179: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.202.51:5060;branch=z9hG4bK14031c57 From: BR2Ph1 sip:3001@10.10.202.1;tag=fcfbfbcb61d6002086bfb221-0332f79c To: sip:3600@10.10.202.1 Date: Mon, 05 Sep 2011 12:50:41 GMT Call-ID: fcfbfbcb-61d60013-326b97fb-35906d9e@10.10.202.51 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 *Sep 5 12:50:41.183: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:3600@10.10.210.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060
Re: [OSL | CCIE_Voice] Problem with UC and CUCME integration
Hi team Can you tell me where I can check the MWI setting in UC whether it is notified or Unsolicited? I tried to configure as unsolicited but it doesnt work. Thanks team --- On Tue, 6/9/11, Ashraf Ayyash ash.ayy...@gmail.com wrote: From: Ashraf Ayyash ash.ayy...@gmail.com Subject: Re: [OSL | CCIE_Voice] Problem with UC and CUCME integration To: Rynard Coetzee rynard.coet...@bytes.co.za Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Received: Tuesday, 6 September, 2011, 12:33 AM glad to hear its working , have a good study/day Ash On Mon, Sep 5, 2011 at 4:47 PM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Ash You were spot on mate ,it was the server on the port group page that was pointing to the incorrect IP ,was pointing to Loopback address where my SIP was bound to previously ,I had changed the binding to the voice vlan but forgot to change it on UC. Working 100% now. Thanks for the help. -Original Message- From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com] Sent: 05 September 2011 03:16 PM To: Rynard Coetzee Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with UC and CUCME integration Hello , the CME is sending 2 invite to the CUC and the CUC havnt accept the call with routing failed issue : Warning: 399 'Routing failed: socket=10.10.210.12:5060' review your config once again , why the CME sending 2 invite with 2 diff ips ? and make sure that the server you added from the port group page in the CUC have the same ip which is the IP of your Vlan400 . Ash On Mon, Sep 5, 2011 at 2:56 PM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: *Sep 5 12:50:41.175: //-1//SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:3600@10.10.202.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.202.51:5060;branch=z9hG4bK14031c57 From: BR2Ph1 sip:3001@10.10.202.1;tag=fcfbfbcb61d6002086bfb221-0332f79c To: sip:3600@10.10.202.1 Call-ID: fcfbfbcb-61d60013-326b97fb-35906d9e@10.10.202.51 Max-Forwards: 70 Date: Mon, 05 Sep 2011 12:50:41 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7961G/8.5.3 Contact: sip:3001@10.10.202.51:5060;transport=udp Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: BR2Ph1 sip:3001@10.10.202.1;party=calling;id-type=subscriber;privacy=off;sc reen=yes Allow-Events: kpml,dialog Content-Length: 276 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 14250 0 IN IP4 10.10.202.51 s=SIP Call t=0 0 m=audio 18866 RTP/AVP 0 8 18 101 c=IN IP4 10.10.202.51 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv *Sep 5 12:50:41.179: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.202.51:5060;branch=z9hG4bK14031c57 From: BR2Ph1 sip:3001@10.10.202.1;tag=fcfbfbcb61d6002086bfb221-0332f79c To: sip:3600@10.10.202.1 Date: Mon, 05 Sep 2011 12:50:41 GMT Call-ID: fcfbfbcb-61d60013-326b97fb-35906d9e@10.10.202.51 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 *Sep 5 12:50:41.183: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:3600@10.10.210.12:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK500414AF Remote-Party-ID: BR2Ph1 sip:3001@10.10.202.1;party=calling;screen=yes;privacy=off From: BR2Ph1 sip:3001@10.10.202.1;tag=4907AFE8-8CD To: sip:3600@10.10.210.12 Date: Mon, 05 Sep 2011 12:50:41 GMT Call-ID: 7EFAA7F8-D6F411E0-960DF543-3ECC855@10.10.202.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2130276264-3606319584-2517103939-0065849429 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1315227041 Contact: sip:3001@10.10.202.1:5060 Call-Info: sip:10.10.202.1:5060;method=NOTIFY;Event=telephone-event;Duration=2000 Expires: 180 Allow-Events: telephone-event Max-Forwards: 69 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 184 v=0 o=CiscoSystemsSIP-GW-UserAgent 727 0 IN IP4 10.10.202.1 s=SIP Call c=IN IP4 10.10.202.1 t=0 0 m=audio 30320 RTP/AVP 0 c=IN IP4 10.10.202.1 a=rtpmap:0 PCMU/8000 a=ptime:20 *Sep 5 12:50:41.199: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 503 Service Unavailable From: BR2Ph1 sip:3001@10.10.202.1;tag=4907AFE8-8CD To: sip:3600@10.10.210.12 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK500414AF Expires: 180 Call-ID: 7EFAA7F8-D6F411E0-960DF543-3ECC855@10.10.202.1 CSeq: 101 INVITE Warning: 399 'Routing failed: socket=10.10.210.12:5060' Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,SUBSCRIBE Content-Length: 0 *Sep 5 12:50:41.203:
[OSL | CCIE_Voice] Route Plan questions
Hi Experts This is not exactly a specific question but rather hoping you can share your experience. I am going through the topic of route plan. From various lab exercises, I can see that there are quite a few different approach or way to achieve the route plan requirements. Then I have to ask, how do you analyse the route plan requirement or how do you decide which way to achieve the requirements? For example, I can see some people use transformation CSS at device pool or some use them at the gateway. Or in terms of translation pattern, some will use one partiton for all sites, or one per each site. I suppose the end result is to make sure it works according to the requirement, I just want to see how everyone tackle this, especially when you only have less than 8 hours to do this. Cheers ___ 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] Stange problem with Subscriber and Publisher
Hi KP Thanks for the advise. The command itself works, but the issue still exists, let me explain. After the command, a message saying the database is successfully replicated. I can see the latest configuration on the phone as they registered to the Sub after it restarted. But I still can not reset the phone which is registered to the Sub, and the new configure doesnt seems to pass to the Sub? Did I miss something here? --- On Tue, 2/8/11, CCIEVoiceKP ccievoic...@gmail.com wrote: From: CCIEVoiceKP ccievoic...@gmail.com Subject: Re: [OSL | CCIE_Voice] Stange problem with Subscriber and Publisher To: Vega Wong vega2...@yahoo.com.au Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Received: Tuesday, 2 August, 2011, 4:19 PM My bad ... utils dbreplication forcedatasyncsub all is what I meant. KP On Mon, Aug 1, 2011 at 8:54 PM, CCIEVoiceKP ccievoic...@gmail.com wrote: I've run into database synch problems in my setup (pub and sub on esxi). SSH into the pub and issue utils dbreplication forcesynch all When it's finished reboot the sub and see how that works. KP Sent from my iPhone and I have big thumbs ... So please excuse the typos. On Aug 1, 2011, at 7:51 PM, Vega Wong vega2...@yahoo.com.au wrote: Hi Experts I am having a strange problem with my Pub and Sub. I have installed them on EXI, and the services can starts up in each of the CUCM fine. My problem is, when the phones are registered to the Sub, any config other than the DN doesnt seems to pass down to the phones. i.e. Date group, Phone button template. I cant even get the phones to reset. But when the phones register to the Pub, (I shutdown the Sub) those configs appear and everything works fine. I have tried using both as TFTP servers, restarted the TFTP services on both. Still the same. Any ideas? ___ 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] Stange problem with Subscriber and Publisher
Hi Experts I am having a strange problem with my Pub and Sub. I have installed them on EXI, and the services can starts up in each of the CUCM fine. My problem is, when the phones are registered to the Sub, any config other than the DN doesnt seems to pass down to the phones. i.e. Date group, Phone button template. I cant even get the phones to reset. But when the phones register to the Pub, (I shutdown the Sub) those configs appear and everything works fine. I have tried using both as TFTP servers, restarted the TFTP services on both. Still the same. Any ideas? ___ 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] Plus sign on display of phone - assistance
Hi In that case, you can use translation rule to add the + sign in front of the number --- On Thu, 28/7/11, CCIE for Me cciefo...@hotmail.com wrote: From: CCIE for Me cciefo...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Plus sign on display of phone - assistance To: Kshitij Singhi martinian.ksin...@gmail.com, ccie_voice@onlinestudylist.com Received: Thursday, 28 July, 2011, 3:20 AM What if the callmanager is not involved, for instance, one site is CME calling another site that is also CME? From: Kshitij Singhi Sent: Wednesday, July 27, 2011 2:30 AM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Plus sign on display of phone - assistance If it is a 7965 phone then that is the default. The + doesn't show up on the main screen, but does show up in the bottom left hand corner when there is an incoming call. Options to work around this: 1. Use the incoming prefix settings on the H.323 GW page based on ISDN plan/type 2. Use calling party transformations to prefix a + On Wed, Jul 27, 2011 at 11:44 AM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Plus sign on display of phone - assistance (CCIE for Me) 2. CCIE Exam version question (fresita mia) 3. Re: CCIE Exam version question (None Ya) 4. Re: CCIE Exam version question (Marko Milivojevic) 5. hw vpn config (Randall Crumm) 6. Re: UCCX system script (Randall Saborio) -- Message: 1 Date: Tue, 26 Jul 2011 15:37:44 -0400 From: CCIE for Me cciefo...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Plus sign on display of phone - assistance Message-ID: blu165-ds198bb3e11c4ae0b710a407b0...@phx.gbl Content-Type: text/plain; charset=iso-8859-1 I am having a very difficult time getting the + to show up in front of the number on the display of the phone. The bottom of the phone screen where the softkeys are will always say From +442077353001 but the display that opens next to the number says (3001) 442077353001. The + always shows going into an MGCP gateway. I know there is a trick to this but I can't seem to figure it out. All of the digits are coming in fine and the debug shows + is being presented. Do I need to add something to any of the dial-peers? If I remember correctly, the dial-peers strip the + from calls. Or is this all normal behavior and nothing I can do about it. thanks for the help. -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110726/256a486b/attachment-0001.html -- Message: 2 Date: Tue, 26 Jul 2011 14:19:01 -0600 From: fresita mia fresita_mi...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CCIE Exam version question Message-ID: snt133-w4307585e5924ff1bc38c028a...@phx.gbl Content-Type: text/plain; charset=iso-8859-1 From: fresita_mi...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: CCIE Exam version question Date: Tue, 26 Jul 2011 12:17:42 -0600 Hello, Does anybody know what lab they are delivering these days? Is it still lab 5? Thanks. -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110726/70118210/attachment-0001.html -- Message: 3 Date: Tue, 26 Jul 2011 17:00:52 -0400 From: None Ya cciev-iw...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE Exam version question Message-ID: col116-w290360090c2b0a348b80d6f4...@phx.gbl Content-Type: text/plain; charset=iso-8859-1 Even if the labs were numbered, discussing the actual question content of the labs in current use is a violation of the NDA. From: fresita_mi...@hotmail.com To: ccie_voice@onlinestudylist.com Date: Tue, 26 Jul 2011 14:19:01 -0600 Subject: [OSL | CCIE_Voice] CCIE Exam version question From: fresita_mi...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: CCIE Exam version question Date: Tue, 26 Jul 2011 12:17:42 -0600 Hello, Does anybody know what lab they are delivering these days? Is it still lab 5? Thanks. ___ For more information regarding industry
Re: [OSL | CCIE_Voice] Presence, CUPS and CUPC
Hi Thanks for the advices, I sort out the offline issue, it was because I didnt setup DNS for CUPC to connect to CUPS. Once thats setup, IM works as well as Presence status. Just one more thing, by manually adding the contact, I couldnt save the phone number of the contact. That means I cant place a call to the contact in CUPC. But it works when I use the phone control (in CUPC) and actually dial the extension. Is this also due to not using ldap? Cheers ___ 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] Presence, CUPS and CUPC
Hi Experts I am working on CUPS and CUPC at the moment, I just have some questions with the setup. So far I have done: I have successfully set up the integration between CUCM and CUPS. As I can see all the users and the SIP trunks automatically appears in CUPS. IPPM works on IP phones, I can send messages between IP Phones I can add contacts using IPPM, or the User page of CUPS. The contacts will shows up in CUPC When I run the system troubleshooter in CUPS, no Red crosses shown. (Except those items I havent configured - LDAP, voicemail) My issue is with the CUPC, I can log in the system using the User ID. I can see the contact added through IPPM or CUPS User page. I can use the CUPC to control the IP phone. However, at the bottom of the CUPC, it always shown as Connected(limited). Also, I cant search the contact within the CUPS.With the contact added, it always shown as offline. I have read that I will need LDAP in order to make the presence status to work in CUPC, is that true? Can i make this work without the LDAP? Please help Cheers ___ 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] Problem with 7961
Hi Experts I think my 7961 phone is faulty, when I plugged it into the switch, all LED are on and stayed on. Nothing come up on the screen. I tried to factory default it by pressing the # key, but it doesnt seems to response. Has anyone encounter something like this? Please help Thnaks ___ 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] Problem with 7961
Thanks all, I think I have left out part of the story about this. I have been using this phone for over 6 months now, it had been good for both CUCM and CUCME, SCCP or SIP. The problem started this morning when I accidentally dropped it on the floor and since then the problem starts. Just as soon as the phone plugged into the switch, all LEDs lid and stay on. Nothing shows on the screen. I am still hoping I didnt break the phone when I dropped it. Please tell me that is not naive thinking. Cheers --- On Sun, 10/7/11, indigoboy indigo...@gmail.com wrote: From: indigoboy indigo...@gmail.com Subject: Re: [OSL | CCIE_Voice] Problem with 7961 To: Bill Lake whl...@gmail.com Cc: Vega Wong vega2...@yahoo.com.au, ccie_voice@onlinestudylist.com Received: Sunday, 10 July, 2011, 11:05 PM I had this problem once where my 7961 and 7970 would simply NOT work when I issued not only the 1234567879*0# sequence, but also the 3491672850*# (to this day, I still don't know what the difference between the two sequence commands are..maybe someone can shed light). I checked and rechecked all settings...other phones would register fine...just not those two. What made it worse was that the other phones were all 7960s. I was convinced my G2 phones were the cause of the problem and that I had bricked them. For whatever reason, I simply changed CUCM images and everything worked. Sometimes the CUCM is corrupt and isn't sending over the correct files or newer files at startup. On Sun, Jul 10, 2011 at 5:02 AM, Bill Lake whl...@gmail.com wrote: Sometimes phones will act like this if the power is not appropriate. Check a known good phone there on that port or try a power brick plugged into the phone to get it working. If power brick and all does not work maybe someone else has a suggestion Here are the power requirements for some phones. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/white_paper_c11-481292.html On Sun, Jul 10, 2011 at 6:21 AM, Vega Wong vega2...@yahoo.com.au wrote: Hi Experts I think my 7961 phone is faulty, when I plugged it into the switch, all LED are on and stayed on. Nothing come up on the screen. I tried to factory default it by pressing the # key, but it doesnt seems to response. Has anyone encounter something like this? Please help Thnaks ___ 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
[OSL | CCIE_Voice] UCCX integration problem
Hi experts I have installed uccx in vmware, the installation seems fine. However, I am having problem with the CUCM integration (which is also installed in vm). I have tried to run Cisco JTAPI Resync, but I keep getting this error: An error has occurred while trying to resynchronize the Cisco JTAPI Client When I checked the Control Center, under the Unified CM Telephony Subsystem, it comes up with Manages JTAPI calls in CRS Engine, jtapi version unknown I have tried to install the Jtapi client in the IPCC server, no success. I have also tried to modified the setting in cet as suggested from the Cisco Support Forum, but still no success. Please help Cheers___ 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] Setting up AA in Unity Connection
Hi guys I need help in setting up AA in Unity Connection. I have setup a new call handlers called AA in Unity Connection with the extension set to 5000 with the setup of the user inputs. In CUCM, I have created a CTI route point with DN 5000, setting the CFA to voice mail. The problem I am having is, whenever a phone in CUCM calls 5000, it goes to the voice mail greeting of the user AA, instead of presenting the user options. I also noticed that the CTI route point in CUCM doesn't registered to anything, but I am not sure whether that is the cause of this problem. Please point me the right direction? Cheers ___ 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] interesting post
hi guys Where did you attempt your lab exam? Cheers --- On Mon, 20/6/11, Cristobal Priego cristobalpri...@gmail.com wrote: From: Cristobal Priego cristobalpri...@gmail.com Subject: Re: [OSL | CCIE_Voice] interesting post To: George Goglidze gogli...@gmail.com Cc: ccie_voice@onlinestudylist.com Received: Monday, 20 June, 2011, 7:01 AM that's why i said it's interesting, i didn't mean anything else 2011/6/19 George Goglidze gogli...@gmail.com ohhh, don't get me started on this one mate... I could say much... too much. On Sun, Jun 19, 2011 at 7:22 PM, Cristobal Priego cristobalpri...@gmail.com wrote: https://supportforums.cisco.com/message/3380695#3380695 what do you think ? ___ 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 -Inline Attachment Follows- ___ 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] Total bandwidth from multiple PVCs on serial interface
Hi I am trying to setup up two frame relay PVCs on a serial interface (WIC-1T). On one PVC, I will have a bandwidth of 1536kbps, and 768kbps on the other one. This gets me thinking that the total bandwidth would be 2304kbps (2.25Mbps). I may be wrong, but I thought on a serial interface - WIC-1T, the bandwidth is 1.544Mbps? Then how does it works with the combined bandwidth from the two PVCs? Please help Cheers Vega ___ 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
