Re: [OSL | CCIE_Voice] Can't dial from CME to CUCM
You should check dialed number analyzer (https://pub-ip/DNA) to check the cucm if it will route the call from the gateway to the dn. In dna use the 'gateway' option, pick the vgw, and enter a dn you're trying to call. If dna says it will route then you need to focus on the cme side, or if it doesn't roite then focus on cucm side. Without seeing more details about your config I will recommend two thing: 1. Run debugs on cme router to ensure you're matching the dial peer and sending to cucm. 2. Check the css that is applied to the vgw and ensure that css has the partition(s) that contain your DNs. -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jul 12, 2014 9:32 AM, Ben John benjoh...@hotmail.com wrote: Hello everyone, I connected my CUCM with my CME using H.323 gateway. I am facing a little bit challenge i can dial from CUCM to CME but i can't dial from CME to CUCM. I am using 4 digits dialing between both sites on the call manager i created RG using the H.323 gateway i created in the callmanager a RL pointing to gateway and a RP using that RL i also created DP,Region,LOC, etc... On the CUCM my RP looks like this: 6. with a predot On the gateway incoming calls is set to 4 When dial 6 + last 4 digits extension on the CME the phone rings. The challenge is that we have one HQ with and two remote sites all of them under on cluster one PUB and two SUBs the HQ and one remote site register with SUB1 as primary, SUB2 secondary and PUB last the other remote site register with SUB2 as pri, SUB1 as sec and PUB last all three site have different DNs and different area code. HQ last 4 digits start with 7 Remote 1 DNs start with 0 Remote 2 DNs start with 2,4,5,8 and also has two separate area code my configuration on the CME looks like this: This dial-peer go to HQ and Remote 1 dial-peer voice 2000 voip description calls to HQ and Remote 1 destination-pattern ^[0,7]...$ session target ipv4:100.260.129.21 dtmf-relay h245-alphanumeric no vad This dial-peer go to Remote 2 dial-peer voice 3000 voip description calls to Remote 2 destination-pattern ^[2,4,5,8]...$ session target ipv4:100.260.152.20 dtmf-relay h245-alphanumeric no vad When someone call from CME to CUCM the calls die. i am not using any translation pattern i think i don't need one Any idea what i am missing here ? Thanks, Ben ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] MOH audio
You're close. The holder (phone that is initiating the hold) dictates the audio file and the holdee (phone being held) uses its configured Moh server. An example would be if user A is a help desk or other agent and has a recorded voice (special offers, website url, general tips, etc) as the moh audio file, but user B is a standard user with music file for MOH. If A places B on hold, then user B should hear the MOH file set on the A phone relevant to the help desk. For your scenarios it would look like this: A: b hears audio3 (network moh from A) streamed from server B (set on phone b) B: b hears audio2 (user moh from A) streamed from server B (set on b phone) C: a hears audio3 (user moh from b) streamed from server a (set on a phone) In another example, if user B pressed transfer when connected to A then user A would hear silence, not moh, before transfer is completed...hopefully you can see why based on the settings you listed. (Hint, take a look at which file A would be looking for from which server.) Hope this helps -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jun 9, 2014 5:40 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: folks, we have following 2 IP phones registered to same CALL Manager. And there is active call between Phone A Phone B Below are the Line Level setting of both Phones -- 1. Phone A User Hold MOH: Audio 2 Network Hold MOH : Audio 3 MRGL : MRGL A -- MRG A --MOH Server A (Audio 1, Audio 2, Audio 3) 2. Phone B User Hold MOH: Audio 3 Network Hold MOH : Audio 4 MRGL : MRGL B -- MRG B --MOH Server B (Audio 2, Audio 3, Audio 4) Scenario and Understanding -- a. When IP Phone A press TRANSFER , Phone B will hear Audio 4 from MOH server B b. When IP Phone A press HOLD , Phone B will hear Audio 3 from MOH server B c. When IP Phone B press HOLD , Phone A will hear Audio 2 from MOH server A DOES my understanding above correct ? Thanks, Karen ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] PSTN incoming /outgoing not working
Make sure the PRI is up on your router as well: Show isdn status L2 should say 'multiple frame established' and if it doesn't you have an issue. It would likely say 'tei assigned' when there's a problem. On an mgcp gw the L3 status should say 'cucm manager'. Troubleshooting guide here: http://www.cisco.com/c/en/us/support/docs/wan/t1-e1-t3-e3/8131-T1-pri.html If the PRI is up and you still have issues then its likely call routing in cucm. Check the css on the pri for inbound and your route patterns/partitions and the line or device cuss on phones for outbound. -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Apr 22, 2014 9:12 AM, vikas wankhede vikaswankh...@gmail.com wrote: Try below debug, it may help, debug voice ccapi inout debug isdn q931 detail debug isdn q921 detail Thanks. On Mon, Apr 21, 2014 at 6:48 PM, Dharambir kumar varma dharambi...@gmail.com wrote: Hi All Last week, I was unable to make ougoing pstn call,even no incoming call. but service provider telling that pstn line was ok.how can I check on cucm, gateway Mgcp/2811 what message it displays in cucm logs. -- Regards, Dharambir Kumar ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] FXO calls heard on multiple lines
Get two analog phones and plug them directly into the pots lines to test. (If you have more than two lines then you'll first need to identify which two lines cause the problem next time it occurs.) If you still hear the conversation between the two lines with the analog phones then your issue is with the physical wiring and/or provider side. It highly unlikely this is any problem on the voip side unless the fxo wic is bad. This problem was common back in the day when high numbers of analog trunks were common. This could be an issue on a punchblock that inadvertently crossed wires or a cut wire somewhere that these two lines are touching. -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Mar 26, 2014 8:49 AM, Mike O'Nan mdona...@gmail.com wrote: Hello all, I have CUCM 8.6 with an MGCP GW at a remote site. User A makes a call with no issues. User B receives or makes a call at the same time that A is on the phone then B can hear A'sx conversation. It doesn't seem to be 2 way in that when they can hear the call, they tell the customer that called in to wait out the conversation or call in again later. I haven't ran into this before and was wondering if anyone had any opinions? Nothing crazy config wise. In CM I have a route group with the FXO ports and a few route patterns. Thanks for any help! ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] 2nd LD call fails
As a test you can modify the mgcp setting in cucm to NOT send the outbound IE. This will confirm whether the Telco is rejecting the call due to invalid IE. I recently worked a similar issue where I saw an invalid IE error message (a rouge prefix on a rp caused the ani to be too long) but the calls were still routed by the telco and connected successfully. In this case the issue was cosmetic and the ani was easily corrected. -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 30, 2014 11:32 AM, Mike O'Nan mdona...@gmail.com wrote: Its a full PRI from a carrier. I noticed that as well I was just hoping it was some config error on my end. This carrier is a pain to work with! Thanks for the input! Original message From: Moataz Date:01/30/2014 10:15 AM (GMT-06:00) To: Mike O'Nan Cc: ccie_voice-boun...@onlinestudylist.com,ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] 2nd LD call fails I can see the release is coming from the PSTN due to invalid information elements Regards, Moataz Tolba On Thursday, 30 January 2014, 18:08, Mike O'Nan mdona...@gmail.com wrote: Pattern is set off net and I fixed the secondary dial tone...still get reorder tone on 2nd LD call. Any ideas from the debugs I provided? On Jan 30, 2014 9:45 AM, Mike O'Nan mdona...@gmail.com wrote: I just noticed in the trace Outside Dial Tone = NO. I have also confirmed the LD pattern is not set for off net. Interesting that when I set to off net it does not give secondary dial tone until the 3rd digit is dialed. I just watched a video yesterday on how to change that but can't remember off the top of my head? On Jan 30, 2014 9:40 AM, Mike O'Nan mdona...@gmail.com wrote: Here are the debugs from the MGCP GW: RTR-02#debug isdn q931 RTR-02#debug ccm-manager backhaul packets Call Manager backhaul packets debugging is on RTR-02# Jan 30 08:19:12.546: cmbh_rcv_callback: -- Receiving backhaul msg for Se0/3/1:23 : | bk_msg_type = DATA_REQ | bk_chan_id (slot:port) = 0:1 | Q.931 length = 41 | Q.931 message type: SETUP | Q.931 message = 0802008E0504038090A21803A98397200200F36C06218137353739700CA13132373035373732383332 Jan 30 08:19:12.546: ISDN Se0/3/1:23 Q931: TX - SETUP pd = 8 callref = 0x008E Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98397 Exclusive, Channel 23 Net Specific Fac i = 0x00F3 Calling Party Number i = 0x2181, '7579' Plan:ISDN, Type:National Called Party Number i = 0xA1, '1270XXX' Characters hidden Plan:ISDN, Type:National Jan 30 08:19:12.578: ISDN Se0/3/1:23 Q931: RX - STATUS pd = 8 callref = 0x808E Cause i = 0x82E4 - Invalid information element contents Call State i = 0x01 Jan 30 08:19:12.578: cmbrl_send_pak: -- Sending backhauled msg for Se0/3/1:23 : | bk_msg_type = DATA_IND | bk_chan_id (slot:port) = 0:1 | Q.931 length = 12 | Q.931 message type: STATUS | Q.931 message = 0802808E7D080282E4140101 Jan 30 08:19:12.638: ISDN Se0/3/1:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x808E Cause i = 0x8295 - Call rejected Jan 30 08:19:12.638: cmbrl_send_pak: -- Sending backhauled msg for Se0/3/1:23 : | bk_msg_type = DATA_IND | bk_chan_id (slot:port) = 0:1 | Q.931 length = 9 | Q.931 message type: RELEASE COMPLETE | Q.931 message = 0802808E5A08028295 Jan 30 08:19:27.486: cmbh_rcv_callback: -- Receiving backhaul msg for Se0/3/1:23 : | bk_msg_type = DATA_REQ | bk_chan_id (slot:port) = 0:1 | Q.931 length = 41 | Q.931 message type: SETUP | Q.931 message = 0802008F0504038090A21803A98397200200F36C06218137353834700CA13138313232353038343038 Jan 30 08:19:27.490: ISDN Se0/3/1:23 Q931: TX - SETUP pd = 8 callref = 0x008F Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98397 Exclusive, Channel 23 Net Specific Fac i = 0x00F3 Calling Party Number i = 0x2181, '7584' Plan:ISDN, Type:National Called Party Number i = 0xA1, '1812XXX' Characters hidden Plan:ISDN, Type:National Jan 30 08:19:27.518: ISDN Se0/3/1:23 Q931: RX - STATUS pd = 8 callref = 0x808F Cause i = 0x82E4 - Invalid information element contents Call State i = 0x01 Jan 30 08:19:27.518: cmbrl_send_pak: -- Sending backhauled msg for Se0/3/1:23 : | bk_msg_type = DATA_IND | bk_chan_id (slot:port) = 0:1 | Q.931
Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html . Regards, --Somphol. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] Changing Sampling rate on CUCM
Look on service Params cm service. Search (ctrl f) for millisecond and you'll see the sampling interval per codec. Most codecs default to 20 ms (g729 and g711 are the relevant ones at 20ms default). -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 16, 2014 2:32 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello All, Is there a possibility to change the sampling rate on CUCM. If so, please let me know where can I find it. Thanks, Viki ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] Route pattern or route list detail
I recommend doing all digit manipulation at route list. Then if you have a question that specifies what the To display should show you do ALSO do digit manipulation at the route pattern. The rp applies to the phone and rlist would be used for the gateway. -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Dec 28, 2013 1:54 AM, Olusegun Oguntuga segunogunt...@gmail.com wrote: Hi there, Please can someone explain where to manipulate called number? For example a local call 9.[2-9]xx, If I predot at route pattern, the calling phone screen displays To 202 If I predot at route list detail, the calling phone displays To 9202 Which is correct please? I understand for teho calls it has to be at route list detail level. How about non-teho calls.? ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] CCIE Lab Prep via VMware
I have been using an older macbook pro with 16gb to run my vms on fusion. (I was running the officially supported 8gb max up until 2 months ago when I got a 16gb kit on amazon for $150) I use the physical ethernet port in bridged mode (ie no nat, vms have layer 2 access to the eth0 port) for all vms and have my physical switch (hq switch) directly connected. All routers are behind the access switch and when I practice lan qos I just use an unused switchport since my pc port is also my server port. On my mac I also have wifi enabled and that has a default route. The phys eth0 port has a static ip (same subnet as my vms) but NO default route. I use the mac os x terminal app (think cmd prompt on a pc) to add static routes to all vm/lab subnets to use the eth0 while general internet traffic uses wifi default route. In this setup I *could* access gui from mac (ffox, safari, chrome) to cucm but latley I have used an xp vm full screen on an external monitor for my mock labs...to avoid nuances in my mac os x (non-ie browsers and terminal) versus the win xp (ie and putty) when you sit the real lab. If you're willing to spend the cash I highly recommend using hard phones (I just use 7961) and your own switch and routers. The reason I used my laptop rather than a beefy server was because I can take the vms with me anywhere and practice the gui sections while at work or travelling. (I could also use softphones to practice using only my laptop, just like some of the ipexpert bls demo videos.) I think newer versions of fusion let you build a loopback interface for the mac but I actually built an rj45 loopback plug for my physical ethernet port (on a mac the static ip on the physical eth port is only up when it is connected to a switch...or my loopback plug). Regarding vm sizing, fusion (and the free vmware player for pc) does NOT allow you to oversubscribe you ram. If you have 6 vms with 2gb vram each you must have 6*2=12gb ram on the mac...right now I can run 6 vms all together: -pub 1.5gb -sub .75gb -cuc 1gb -ccx 2gb (it doesn't work right for me after change ram post install of app) -cups 1gb -win xp 2gb When I had 8gb previously I could run them all but it was near 100% ram usage on the mac and it slowed to a crawl so I would only run one of the apps (cuc, ccx, or cups) at a time or even shutdown the sub if needed. Bottom line is practice whatever way works for you within your budget. I needed to be mobile so I used my laptop. If I had a dedicated rack and phones at work I would have used a server there instead and just vpn/rdp while remote. On windows laptop vmware player would be very similar - 8gb is do-able, but 16gb is ideal. I think the paid version of vmware workstation (under 100 I think) may let you oversuvscribe ram. If using a random server with esxi (or esxi on a vm inside fusion or player), you can oversubscribe ram but I wouldn't recommend it. Either way you connect your esxi server or laptop to a physical switch. Someone else suggested using the 3750 poe for all phones and skipping the esw wic and nm modules-I agree those aren't anything fancy just use the 3750 and config site b and c as trunk ports instead if avvess ports. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Dec 12, 2013 7:49 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi Actually I mean about switch and router *From:* Bennajer Isidro [mailto:bennajer.isi...@outlook.com] *Sent:* Thursday, December 12, 2013 2:38 PM *To:* Chrysostomos Christofi *Cc:* wilson.sam...@bt.com; kstap...@cisco.com; josh.pe...@gmail.com; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] CCIE Lab Prep via VMware I just started my voice study. Currently im running all my equipment in virtual environment. To install all uc server. U can try to dl vmplayer then install esxi inside of it. On 12 Dec, 2013, at 20:15, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi Could you pls advice how to run a switch and router into vm workstation? *From:* ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.com] *On Behalf Of *wilson.sam...@bt.com *Sent:* Thursday, December 12, 2013 1:09 PM *To:* kstap...@cisco.com; josh.pe...@gmail.com *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] CCIE Lab Prep via VMware Hi Kenneth, I have used VMWare Workstation ver 8 ( I am not very sure what version I am running for the last 1 year) but VMWare Workstation works fine, no issues at all. However, there is one catch, and that is, one is at the mercy of the Host OS's resource allocations (CPUs, Memory etc) and that tends to slow down the installation in most of the times (I have an old Dual Xeon, 16 GB Mem with 500G + 1 TB HDD on which I run these beasts) the Host OS is Win XP 64 Bit and it works, though I can run almost 5 VMs in parallel,
Re: [OSL | CCIE_Voice] CCIE Lab Prep via VMware
No problem, i have a pretty elaborate setup at home and I only included the relevant lab stuff. I actually have gear in my basement so I'm not next to the noise and have multiple vlans on my home network to trunk lab traffic around. On other disclaimer - I have a lot of background deploying/supporting ESXi in a production environment so I'm very familiar with virtual hardware, software, networking, etc. For folks that do NOT have a similar professional background with virtualization my advice would be to keep it simple - use a spare laptop/desktop/server and run esxi and be done with it. My laptop setup took a while to get fine tuned and have only had it setup this way in the last 6 months of studying (the first 9 months i had gear in my office, some vms on mac, some on external ESXi, it was a mess and too loud). Fusion/player isn't too hard in itself and is simplified by not trying to connect multiple networks like I did with my home wifi and lab concurrently - if you can't get it working just turn off your wifi, hard wire into the switch and assidn a static IP (or build a dhcp scope on hq router or switch) - don't worry about internet while you're labbing. I actually just passed my lab this week! I will be migrating to the collab using the written and haven't given much thought about the new lab (aka voice v4 using CUCM 9 or 10 with some video). For those writing off the current voice blueprint (due to no more seats) I would recommend getting at least one ISR G2 with PVDM3 and 3 9971 phones. The major difference here is that the PVDM3 chips can actually be used as a video conference bridge, for example 3 9971 phones. Also get familiar with current IOS like 15.1M/T and 15.2M/T, although there are tons of new features I don't know how many are voice/video related (there's now a trust list for h323/sip dial-peers, by default no IPs are trusted). Video conf reference for ISR G2: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps4952/qa_c67-649850.html The collab blueprint doesn't include the TelePresence products (VCS, TMS, etc) so I would expect that configure a SIP trunk to a cloud video controller (aka VCS) would be fair game. The good news is this is very similar to build a cloud SIP trunk right now on current blueprint - build the trunk and the router group/list/pattern and you're done. bandwidth and codecs are controlled thru locations/regions just like with audio. Time will tell but I would estimate the at least 75% of knowledge gained from studying voice v3 will translate to collab v1 - no one knows *exactly* what will be dropped (frame relay?) and what will be added (video phones). I'm not sure if IPexpert has materials for collab at the moment, but I thought I saw holiday sales on their training that said if you buy voice v3 then you will get collab v1 materials for no additional cost. I've been stuck in CUCM 7.0 for so long I have been avoiding CUCM 9, but now I'm looking forward to it. Do a google search for new features in cucm 9 and consider whether those would be potential new test questions. The big one I would expect is enhanced-locations-CAC and possibly native CUCM call queuing. I would expect CCX to stick around, but native call queuing could likely replace (or compliment) B-ACD. I have learned a lot from this study group and will keep an eye on it for a while to help everyone still pressing on towards CCIE Voice/Collab. It was a long journey for me and this study group definitely on several occaisions - I plan to pay it forward and I hope the rest of you that WILL PASS will do the same for future candidates. Good luck! On Thu, Dec 12, 2013 at 8:34 AM, wilson.sam...@bt.com wrote: Thanks Justin for the post. I have had used the Dell Precision 390 with 16 G RAM, however the noise and the power consumption is just bit crazy and the second most important aspect is that, one can be mobile with Mac Pro, not a doable situation with fat cat server. Btw, have you already passed your Lab? If you are in case preparing for the Collab version, I wanted to know if there are IPExpert is already providing the training for it and what about the lab network layout for the same. Regards Sam Wilson -- *From:* ccie_voice-boun...@onlinestudylist.com [ ccie_voice-boun...@onlinestudylist.com] On Behalf Of Justin Carney [ justin.s.car...@gmail.com] *Sent:* Thursday, December 12, 2013 8:24 AM *To:* Chrysostomos Christofi *Cc:* ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com) *Subject:* Re: [OSL | CCIE_Voice] CCIE Lab Prep via VMware I have been using an older macbook pro with 16gb to run my vms on fusion. (I was running the officially supported 8gb max up until 2 months ago when I got a 16gb kit on amazon for $150) I use the physical ethernet port in bridged mode (ie no nat, vms have layer 2 access to the eth0 port) for all vms and have my physical switch (hq switch) directly connected
Re: [OSL | CCIE_Voice] CUPC Voicemail MWI
You could try leaving the message for the phone w/ cupc, then on the phone dial the DN of the MWI-off (eg, 1999 in IPexpert labs). I'm not sure if dialing the MWI off/on number on the phones even touches the CUC server, but it's possible - if it does sync to CUC in this case then the CUPC MWI would go off and that wouldn't solve your question. This may not be the ideal solution, even if it works, as I would expect that if you powered off the whole system/rack and then back on that the phone/CUC would sync the MWI status (still unread on CUC) and both the phone and CUPC would then show the MWI light. I don't know whether the pod racks are power cycled before grading and I'm not going to hope that they don't power cycle as a strategy. Alternatively, if you need to show a message and it's not explicitly stated as a voicemail or a MWI then you may be able to meet the requirement by showing a new IM as a message from the IPPM phone service to the CUPC user. HTH... Justin On Sat, Dec 7, 2013 at 5:47 PM, Barrera, Hugo hugo.barr...@nexusis.comwrote: Hi, If all MWI’s need to be left off and one of my ip phones is integrated with CUPC and the CUPC client needs to show a message indication what is the trick? *Thanks * ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] SIP Prove TCP is used and Early Offer:- Can any one help in this matter
All 3 of your options are technocally correct for both questions, however I don't know if the lab grading has a preference for which one if you have to indicate the line within the trace using arrows I would personally use the first line of the header for tcp (your option 1) and either the content type sdp (your option 2) if my explanation was vague saying sdp in invite shows eo or i would use the m line of the sdp itself if saying these codec (s) are offered in the invite showing eo. I would not use content length because the other two are more specific, just my opinion (hopefully this works out in my next lab). -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Nov 13, 2013 12:08 AM, IE Target myfrnd...@gmail.com wrote: What proves TCP 1)Outgoing SIP TCP or 2)transport = tcp or 3) VIA What proves Early Offer 1)content-length or 2)content type or 3)SDP itself Outgoing SIP TCP message to 157.26.1.253 on port 5060 index 1 INVITE sip:321234567890@157.26.1.253:5060 SIP/2.0 Date: Tue, 12 Nov 2013 18:36:17 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY From: HQ Phone 1 sip:2001@142.100.64.12 ;tag=2af0b7a6-cadd-470e-807f-59a95570f443-34173098 Allow-Events: presence, kpml P-Asserted-Identity: HQ Phone 1 sip:2001@142.100.64.12 Supported: timer,resource-priority,replaces Min-SE: 1800 Remote-Party-ID: HQ Phone 1 sip:2001@142.100.64.12 ;party=calling;screen=yes;privacy=off Content-Length: 214 User-Agent: Cisco-CUCM7.0 To: sip:321234567890@157.26.1.253 Contact: sip:2001@142.100.64.12:5060;transport=tcp Expires: 180 Content-Type: application/sdp Call-ID: 4fbd2c00-28217521-1-c40648e@142.100.64.12 Via: SIP/2.0/TCP 142.100.64.12:5060;branch=z9hG4bK05cf14a0d CSeq: 101 INVITE Session-Expires: 1800 Max-Forwards: 70 v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 142.100.64.12 s=SIP Call c=IN IP4 142.100.64.12 t=0 0 m=audio 24578 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] XML Services
First, make sure you restart (or even reset) your phone after making a change to directories. Seeing XML file from a browser is a good start, I'm assuming that's what you mean by can reach the XML file... The syntax looks OK offhand, but I always pull up the default enterprise parameters URL and point a browser there for the sample syntax. If you have a permissions issue then create a new XML file on CCX and save it DIRECTLY in the inetpub folder (ie, don't save on desktop and move to inetpub folder). I always remove the leading/trailing whitespaces in the XML (i see your lines are indented) but I don't believe that actually matters. Try doing that and creating a new file just to test. File name should end with .xml not .txt. There a few things to consider when using both for services provisioning. First, this means phones will use both Internal and External services as the name implies. Internal services are the built-in services when CUCM is installed (missed, received, placed, personal, corp, AND VOICEMAIL) and external are specified by URL. I believe the external services are listed first, then the internal are listed second (please test - it could be the other way around). Like most settings in enterprise/service parameters these apply to all devices, UNLESS a setting is applied at the phone level. If you have the enterprise services URL and also a URL on a phone, the one on the phone is used for external plus any/all of the internal services. In this case the enterprise params url is ignored. This could be useful if you want to maintain 'default' services for all phones but one you can set the URL on just one phone. If you want to change all phones you can set the ent param for all phones and then optionally override that with a different custom URL on one phone. I have seen different strategies on these directory questions. Personally I do not delete the default internal services, instead I just disable them if I don't want to want them there and need to use both For example one of the IPexpert practice labs says disable all services but voicemail on a 'lobby' phone and don't affect other phones. Voicemail can ONLY be an internal service. The solution here it to mimic the full set of default internal services by creating an XML on contact center and put this URL in ent params, setting the global services provisioning to 'both. All internal services except voicmail should be disabled, so all phones will get missed/placed/etc. from the URL and VM from internal. For the one lobby phone simply set the services provisioning to internal (it will ignore the ent param URL since that is external and the phone only looks for internal). The only internal service is voicemail and you have met the requirements for this question. Hope this helps... - Justin On Fri, Nov 8, 2013 at 6:14 AM, Nicolas MICHEL mcl.nico...@gmail.comwrote: Hey guys I m using this xml for my services, just to check the structure. Unfortunately when I press the service button nothing appears on the screen of my 7961 ... I am pretty sure I am doing everything right here. Services provisionning has been set to : BOTH and URL has been configured also. I can reach the XML file on the UCCX but still not working ... Any ideas ? Thanks NIcolas PS : XML file CiscoIPPhoneMenu MenuItem NameMissed Calls/Name URLApplication:Cisco/MissedCalls/URL /MenuItem /CiscoIPPhoneMenu ___ 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] H323 gateway doesn't send the plus on outgoing PRI calls
If you're adding the plus I cucm this is expected behavior that it will be lost at the ios gw. You're not missing anything on cucm, this is just how h323 works on ios routers...the only option is to add a plus in the ios gateway. IOS/h323 will (by design, unfortunately) discard a plus in dnis on the inbound call leg, so make sure you are adding it on the outbound dial or voice port. (You might be able to add plus on an inbound dial peer but I haven't tried and not sure if it would then get discarded.) For the lab I would recommend doing ALL h323 digit manipilation on the h323 gw. If you do it in cucm then later when you need to do srst you will need extra dialpeers and/or translations. For example, I use only 4 outbound pots dialpeers in h323 each with a translation profile that modifies ani (for each site, matching 2... 3... and 4...) and dnis (usually just type/plan). I send all the dialed digits from a phone to the h323 gw the let the gw make the ani and dnis match the pstn requirements for all call types including teho. The only time I modify digits in cucm is when no new rp is alowed for 911 at site b and that uses slrg - the ani is masked to 7digits in cucm, but the h323 still must set ani/dnis type and plan to unknown for dnis. Also, for teho from site a to site b pstn, I would strip the 91408 from the dialed digits and prefix 9 in the route list/rg for h323 so it matches my outbound local dialpeer. Hope this helps. -Justin On Nov 4, 2013 2:19 AM, Paul Onwude ponw...@gmail.com wrote: *Hi All. * *Need expert opinion on this.* *I have a H323 gateway and i have setup called party transformations on CUCM to send “+” to the gateway. My issue is i don’t see the plus when i debug ids q931. When i do other digit manipulation like adding “#”, it show up on the gateway but not the “+”* *I know i can probably achieve this using translation on the GW but i can’t help but think there is something i am missing in CUCM.* *Any ideas??* *Paul* ___ 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] AAR Configuration
The aar-group setting on device pool does NOT get pushed to all devices in the device pool, while the aar-css does. My strategy is to set aar-css at the dev pool and to manually set the aar-group on EVERY device (phone, gw, vm or cti port, etc) and EVERY line/dn. This take no thought while provisioning thing and when I get to an aar question the only thing to build is the rlist (maybe, if an existing doesnt match exactly) and route pattern. That said my strategy is slightly overprovisioned to save time. I did thorough testing and came up with the minimum config for aar: 1. The calling entity must have the AAR-CSS on the DEVICE (there is no aar css field on a dn, it only exists on a device/port/gw) 2. The calling entity must have the AAR-GROUP set on Either Device *OR* Line/DN 3. The called/target LINE/DN must have the AAR-GROUP. (this makes sense, as you call a dn and you don't care which device(s) have a line appearance for this dn.) if the called DN doesn't have the aar-group it will NOT work, regarless of whether the device where the dn is assigned has the aar-group In summary, my strategy pit the group every where and the css on devices and I don't have to memorize the minimum req in the lab - or more importantly I don't revisit config pahes just to setup aar. Hope this helps... -Justin On Nov 4, 2013 6:57 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi Guys, i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and AAR-CSS on device pool it does not take effect rather i have to apply it each phone device and GW inorder for it to work.Is there any thing i am missing 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
[OSL | CCIE_Voice] how-to: globalize/localize on missed call list vs display
Study group – below is a how-to/strategy response I sent to a friend earlier today who asked about the globalization/localization questions in the practice labs. I'm sharing here in case anyone else needs some pointers and has not yet asked the study list. I hope this helps someone…and if not that's fine, it already helped the person I wrote it for :-) *QUESTION*: CngPTP setup - globalize/localize on missed call list vs display. Should I globalize on the GW according to NPI/TON or do it manually using a CngPTP CSS? *ANSWER*: To solve those questions about modifying the display of a phone it helps me to break it into two steps/phases, at least conceptually. Once you practice it a few times you can configure it in one shot in a specific order, but don't try to do that until you nail the config down and do it a few times. *Phase 1*: IGNORE the phone display, focus on globalization - globalize the inbound call, make it show in the call list, and make it routable when you call outbound from the missed call list. 1. Look at the inbound q931 from the gw to see the PSTN's ANI, make note of the TON. Go to the gateway config in CUCM (either mgcp or h323), and set the prefix based on TON, for example if subscriber with 10 digits then set the sub prefix to +1 or whatever you need. Make this work first (i.e., make it show on the phone), then move on. 2. After you get a missed call with + in the call log (and for now + on the display), make it routable when redialing. Copy/create a RLIST (this is important for the display edits later, DON'T reuse an existing route list), and do your DDI/TON on the RLIST level to make the + call work – only worry about the specific source phone and destination number listed in the question. When this call works (ensuring the PSTN requirements are still met from the routing section), move on to the next phase. *Phase 2*: Now fix the localized display on the phone. 1. INBOUND – you have already modified inbound ANI on the gw (in CUCM) to be the +e164 number. Now create a new CngPTP (in a new PT or a placeholder xform-partition) to match this + number and manipulate to what the question states. The CSS that contains the PT for the CngPTP (either new CSS or a placeholder xform-css which contains form-partition) should now be applied at the target PHONE, remember to UNCHECK 'use dev pool CSS.' You should now have localized the inbound connected number and still have + in call log. 2. OUTBOUND – you should already be able to place the return call from call log, using RLIST for digit manip (part of Phase 1). The reason to do digit manip on RLIST is so that here you can ALSO do digit manip on the Route Pattern. The RP digit manip will be used for display on phone only because RLIST also does digit manip and RLIST will override the RP. (Side note, if you do DDI on RP and NOT on RLIST, then the DDI RP will take effect for the DNIS. In my labs I always do DDI RLIST except special situations like this or when I skip a RLIST and have the RP point directly to an ITSP trunk (and if the display number doesn't matter).) 1. Note – depending on which digits you are removing on the RLIST to make the call route (from Phase 1) the dot may be in the wrong place for your use pre dot strip here in the RP. In this case, move the dot to the right so you can use DDI for phone display to localize, but make sure to go edit the RLIST to prefix those extra digits that the gw needs to keep. 2. Example – you may have originally used +44.2077961234 and predot in RLIST to make the call route. Now for localization you may need to display 77961234 so you need +4420.77961234 in the RP and predot here. Since you moved the dot the DDI in the RLIST is broken, so now modify the RLIST based on the new dot position to prefix the 20. Now that it makes conceptual sense, practice a few times on different sites – make up your own questions/requirements once you get bored with the practice lab question. After a couple times, you'll see the 'big picture' when you read the question and you can go through the whole process and setup all the DDI on the first shot at the RL and RP (rather than doing the RL, then changing it after you realize the RP ddi needs to move the dot for phone display). Good Luck! ___ 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] 4ESW modules QoS/Marking/trusting
The esw modules do not have any qos features. There's no need to use access lists because traffic entering the esw will not be remarked - the default (and only) l2 option is to trust the markings received from the switch ports. For lan qos focus on the 3750. On a slightly related note there are other things to mark on site b and c routers you can use ip qos dscp cs3 sig inside voip dial peers, prefix mgcp to that for mgcp signalling, and use sccp ip precedence 3 for sccp traffic. That should handle signaling markings from router-generated traffic (default for these is af31). Other option is to use acl/nbar to mark traffic. On Oct 29, 2013 1:01 PM, StefanoS stefan...@gmail.com wrote: ...in other words to trust or not to trust incoming traffic from phones connected to the 4ESW modules. I think to be at the safe side we should use access lists. But there's more configuration to be done (not if you use auto qos and FRF.12 LFI) Thanks, Stefanos On Tue, Oct 29, 2013 at 6:47 PM, StefanoS stefan...@gmail.com wrote: Hello all. What do you think is the best practice approach in the lab exam? Using mls qos trust commands on the 4ESW phone ports where the phones are connected (trusted devices) or use access lists for classification and marking and then applying accordingly to the policy? Thanks you, Stefanos ___ 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] which route pattern discard digits includes even # dialing
Nice! I wonder why training material doesn't use that method, I just assumed two patterns was the best/only way and never thought about trying to combine them. On Oct 12, 2013 9:13 AM, William Bell b...@ucguerrilla.com wrote: Actually, you could use the pattern 9.011![0-9#] to cover both dialing scenarios with one pattern. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 11, 2013, at 9:42 PM, Justin Carney wrote: You need both patterns. The first step is matching a route, then digit manipulation is applied. The two patterns are used to match international calls with variable length digits both with and without dialing #. You need the one with # when the question states something like give users the ability to avoid interdigit timeout. This pattern will only match when user dials the # and you could use predot trailing # for ddi. The pattern without # will only match if a user does not dial # and t302 timer expires. The only time you can get away with only one pattern is if the question says you do NOT need to give users a way to avoid interdigit timeout. My strategy is to always use both patterns unless the question says prevent users from avoiding interdigit timeout in which case this extra config with the # pattern would cause you to lose those points. On Oct 11, 2013 12:59 PM, virajith vir...@rediffmail.com wrote: Hello, I wanted to know which discard digits option in route pattern includes both 9011.! and 9011!# dialing . So that only 1 route pattern is created instead of 2 for dialing without and with #. -Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Get your own *FREE* website, *FREE* domain *FREE* mobile app with Company email. *Know More *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-1-10-13___cmp=hostlnk=sign-1-10-13nsrv1=host ___ 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] Voice Lab routing issue
Without seeing your config it sounds like a vlan mismatch between switch and router. Make sure your switchport facing the router is trunking and the router has subinterfaces with the correct vlan tag. Native vlan number will also need to match. If you post those relevant parts of the config I can review. On Oct 11, 2013 5:15 PM, Dane Warner dwar...@epochuniversal.com wrote: All, ** ** I’ve put together my voice lab and I’m having a routing issue which is preventing me from completing the setup. ** ** From SiteA-rtr, I cannot ping 10.10.100.3 on the 3750. The switch cannot ping 10.10.100.1 on the router. I have the PSTN-WAN router connected to f1/0/24 on the switch. From PSTN router I can ping 10.10.100.3 but not 10.10.100.1. The servers can all talk to each other, and the SiteA router can ping the servers, but the servers cannot ping 10.10.100.2 (NTP). It seems to be the connection from Switch port 1 to SiteA router subinterface for VLAN 101 only. The initial configurations came directly from Proctorlabs and I’ve edited from there. Can anyone take a look and see what I’m missing. It must be something obvious but I’m not seeing it. ** ** Thanks much, ** ** ** ** *Dane Warner, CCVP* *Sr. Network Engineer* *Epoch Universal, Inc.* *(909)226-0755* *dwar...@epochuniversal.com * *[image: Epoch_Logo_Smaller_Transparent]* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Karen Johnson *Sent:* Monday, October 07, 2013 10:20 AM *To:* Ramcharan Arya; Josh Petro *Cc:* ccie_voice@onlinestudylist.com; sanity insanity *Subject:* [OSL | CCIE_Voice] ccie written ** ** hi Arya and all, when you have 2 ccie specialization, do you need to write WRITTEN exam for both or just one ? K ** ** *From:* Ramcharan Arya ramcharan.a...@gmail.com *To:* Josh Petro josh.pe...@gmail.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; sanity insanity networksanitytoinsan...@gmail.com *Sent:* Wednesday, October 2, 2013 6:59:08 PM *Subject:* Re: [OSL | CCIE_Voice] Presence - on hook and off hook status** ** ** ** Hi Josh, I do not believe it is related to vmware environment. I am assuming your CUPS is integrated with CUCM using SIP trunk. Can you enable SIP debug level to detail and run collect SIP debug logs ( on primary call processing engine i.e. Sub) and check SIP logs why there is delay in status.? Im my home lab I never had this issue it works almost instantly my CUPS client is installed on UCCX server. Regards, Ramcharan Arya CCIE # 28926 ( Voice/RS) ** ** On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro josh.pe...@gmail.com wrote:** ** I have a huge delay in my presence updates on my system. Im assuming thatapos;s from CUPS being installed in my lab vmware environment though. Try to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone knows how to fix the lag, please let me know. Im assuming its related to vmware. Josh On Oct 1, 2013 11:20 AM, Ovidiu Popa ovi.p...@gmail.com wrote: Hi ** ** You could try a reboot of the CUPS server. Worked for me a couple of times... ** ** Cheers, Ovidiu ** ** On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan martinsloa...@gmail.com wrote: Hi MJ, Is the end user assigned on the line level of the hard phone? That assignment is unique per line appearance so if you make the association on the CUPC device it does not automatically populate to the hard phone/any other line appearance. When the phone goes off-hook CUCM checks the end user assignment for that appearance and if there is an end user assigned it check whether that end user is assigned CUP licensing to decide if the publish message is sent over the CUPS SIP trunk. BR, Marty ** ** On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hello all, I have configured presence and both softphone and deskphone modes , IM and voicemail is working fine on the clients However I have a question when I lift the handset of the phone ( hard phone ) that is assoicated with the CUPC clients . I see that the presence status does not show On the phone and does not turn yellow. I have tried reseting my sip trunk pointing to the presence server yet I see the same issue. Please let me know what can be done to fix this ? Also is this a major issue ? -MJ ** ** ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ** ** ___ For more information
Re: [OSL | CCIE_Voice] TEI_ASSIGNED
If all servers are online you should be registered to you primary server, which in the IPexpert practice labs will be your sub. You show this server as down and are registered to the backup, which I would presume is your pub. Your config (from an earlier email) mgcp call-agent 177.1.10.20 service-type mgcp version 0.1you should be registered here (Primary) ccm-manager redundant-host 177.1.10.10you are registered here (first backup) -you don't have a third cucm, or second backup configured so that will always be none on sho ccm You need to troubleshoot why the gateway is not registering to the sub (your primary ccm). 1. is the ccm service activated and running on the sub? (may need to restart the service) 2. is your db replication good? (use reporting page or cli commands to check 3. to fix db you can try utils dbreplication repair all which is not the most intrusive option but often can clear up issues. beyond this lookup cisco docs on all options/methods to check/fix db replication issues 4. can you register phones to the sub? if no, you likely have db replication issues. Also, since you were able to get your gateway registered now, can you share what you did to fix it with the study group mailer so that others may benefit from what you learned? I provided a lot of recommendations and it would be helpful to understand which were helpful and applied to the error messages you were getting. Thanks, Justin On Sat, Oct 5, 2013 at 11:59 PM, Anthony Nwachukwu anwachu...@accesspointafrica.com wrote: Looks ok now the only problem now is the primary is down should be standby ** ** Why CorpHQ#show ccm-manager MGCP Domain Name: CorpHQ.ccievoice.com PriorityStatus Host Primary Down 177.1.10.20 First BackupRegistered 177.1.10.10 Second Backup None ** ** Current active Call Manager:177.1.10.10 Backhaul/Redundant link port: 2428 Failover Interval: 30 seconds Keepalive Interval: 15 seconds Last keepalive sent:13:57:12 PDT Oct 5 2013 (elapsed time: 00:00:01) Last MGCP traffic time: 13:57:12 PDT Oct 5 2013 (elapsed time: 00:00:01) Last failover time: 13:57:12 PDT Oct 5 2013 from (177.1.10.20) Last switchback time: 13:49:47 PDT Oct 5 2013 from (177.1.10.10) Switchback mode:Immediate MGCP Fallback mode: Not Selected Last MGCP Fallback start time: None Last MGCP Fallback end time:None MGCP Download Tones:Disabled TFTP retry count to shut Ports: 2 ** ** Backhaul Link info: Link Protocol: TCP Remote Port Number: 2428 Remote IP Address: 177.1.10.10 Current Link State: OPEN Statistics: Packets recvd: 1 Recv failures: 0 Packets xmitted: 1 Xmit failures: 0 PRI Ports being backhauled: Slot 0, VIC 0, port 0 Configuration Auto-Download Information === No configurations downloaded Current state: Waiting for commands Configuration Download statistics: Download Attempted : 6 Download Successful : 0 Download Failed : 6 TFTP Download Failed : 33 Configuration Attempted: 0 Configuration Successful : 0 Configuration Failed(Parsing): 0 Configuration Failed(config) : 0 Last config download command: ** ** *From:* Justin Carney [mailto:justin.s.car...@gmail.com] *Sent:* 05 October 2013 20:31 *To:* Anthony Nwachukwu *Cc:* Anthony Nwachukwu; ccie_voice@onlinestudylist.com, ( ccie_voice@onlinestudylist.com) *Subject:* RE: [OSL | CCIE_Voice] TEI_ASSIGNED ** ** You will not be able to bring up the pri until your mgcp gw registers to cucm. In my very first reply I assumed it was already registered but I guess I should have asked. The most common reason for not registering is a mismatch on hostname. The gateway will register as either hostname (if ip domain-name not set) or hostname.domain.com (if ip domain-name domain.com). I see your hostname is CorpHQ and I didn't see the ip domain-name command in yiur posted config. In this case make sure that in cucm you list the mgcp gw as CorpHQ exactly (I always type in the correct/matching case, I'm not sure if it is actually case sensitive but why take that chance?) Use show inventory on the router and make sure cucm has the correct router model and nm/wic part numbers. If still no luck (just looking at cucm), then delete
Re: [OSL | CCIE_Voice] Gatekeeper
In your HQ GK config you have not specified the IP address to which the GK will be bound, commented below in blue: gatekeeper zone local HQ cisco.com NO IP specified here, router will pick one (see below) no zone subnet HQ default enable zone subnet HQ 10.1.5.3/32 enable zone subnet HQ 10.1.5.2/32 enable zone subnet HQ 10.1.130.1/32 enable no shutdown ! If you do not specify an IP on the zone local command the router will pick a specific IP - I don't recall the exact rule offhand, but it may be the highest numbered loopback, and if none the highest physical interface. That default method doesn't matter since you can (ie, should) manually assign the specific IP you want to use and not worry about how the router will pick if you don't. *FIX - Assign an IP for the GK to listen on:* gatekeeper zone local HQ cisco.com *10.1.110.1 * specify the desired IP here after the domain - you can optionally specify a port after the IP, but if you don't it will default to1719 (and will be listed in show run with port 1719), which is fine unless the question tells you otherwise You should probably shut/no shut your gatekeeper after this change. It may even require you to remove the existing zone local HQ first. To speed up your BR2 gateway re-registering do a no gateway then gateway on that side. When I lab I always specific the GK IP even if not called out by the question. I typically use the loopback0 for GK and if using a CUBE I typically will use the voice vlan SVI - but it doesn't matter unless the question states what to use. For reference, the other related commands commented below: *! BR2 side* interface Vlan130 ip address 10.1.130.1 255.255.255.0 h323-gateway voip interface specifies to use this interface to source RAS h323-gateway voip id HQ ipaddr 10.1.5.1 1719register to zone HQ at gatekeeper with IP 10.1.5.1 on port 1719 h323-gateway voip h323-id BR2the local gateway's h323 identifier is BR2 h323-gateway voip tech-prefix 56tell the GK that my tech-prefix (to get to me, BR2) is 56 (on the GK you will see tech prefix 56*) ^make sure the above BR2 commands are applied on the interface you want BR2 to source, and the IP called out on the HQ side is listed on the zone local *! HQ side* interface GigabitEthernet0/0.110 encapsulation dot1Q 110 ip address 10.1.110.1 255.255.255.0 ip helper-address 10.1.5.2 h323-gateway voip interfaceuse this interface to source RAS messages (when talking TO another GK, not when you ARE the GK...when you ARE the GK that is the zone local IP indicated above). When setting up a CUBE on the same router as the GK then this command will determine which IP the CUBE registers to the GK with even though they are the same router. Both addresses can even be the same, i believe. h323-gateway voip bind srcaddr 10.1.110.1use this interface to source h.225/h.245/rtp traffic. If your question states something like the phone's IP address should not be know by the cloud gatekeeper and should only see media from IP w.x.y.z then this command is used to pin media (along with media flow through on the voip dial peer) to the desired IP. Otherwise, this IP needs to match on the CUCM side if the gateway is using h323 to CUCM instead of mgcp. ^make sure the above HQ commands are applied on the interface you want the HQ router to source for CUBE, they are unnecessary (and don't do anything) for the GK process and are needed for cube (or if the gateway is h.323 then the voip bind src is needed to match the configured IP in CUCM) Hope this helps... -Justin On Mon, Oct 7, 2013 at 8:56 PM, Josh Petro josh.pe...@gmail.com wrote: Hi All, I have a strange issue I ran into on a lab recently. The BR2 gateway would not register to the HQ gatekeeper unless I changed the IP address from the 'voice' subnet IP to the 'data' subnet IP. The question said I could not configure the gatekeeper with Zone Prefixes, Aliases nor could I register any e.164 addresses with it. It also said I could only allow the CUCM and BR2 endpoints to register to it. That basically left me to use the Zone Subnet commands. Why would the BR2 gateway not register until I changed the command on the VLAN interface from this: interface Vlan130 ip address 10.1.130.1 255.255.255.0 h323-gateway voip interface * h323-gateway voip id HQ ipaddr 10.1.110.1 1719 G0/0.110 interface* h323-gateway voip h323-id BR2 h323-gateway voip tech-prefix 56 to this interface Vlan130 ip address 10.1.130.1 255.255.255.0 h323-gateway voip interface * h323-gateway voip id HQ ipaddr 10.1.5.1 1719 !gig0/0.5 interface* h323-gateway voip h323-id BR2 h323-gateway voip tech-prefix 56 Here's the config HQ interface GigabitEthernet0/0 no ip address duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/0.5 encapsulation dot1Q 5 ip address 10.1.5.1 255.255.255.0 ! interface GigabitEthernet0/0.10 encapsulation dot1Q 10 ip
Re: [OSL | CCIE_Voice] TEI_ASSIGNED
Did you try the steps I outlined earlier? Does show ccm indicate your mgcp gw is regisyered to cucm? If not check your hostname and domain name (if set) on gw and double check the cucm config. If so it is registered you could take it a step further and remove the pri-group from the controller (and essentially all pri and mgcp config) reboot, then put it all back. Also, if this is your home lab (or if you have access) what does the pstn side of the pri config look like? Make both sides have the same isdn switch-type and the pstn side has isdn protocol-emulate network under serial interface. A reload of both routers wouldn't hurt either (before or after all the above). Some links that may help: http://goo.gl/QA6qaC http://goo.gl/1ctXWS On Oct 5, 2013 10:03 AM, Anthony Nwachukwu anwachu...@apafrica.com wrote: CorpHQ#CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE CorpHQ#CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE CorpHQ#CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE CorpHQ#CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE CorpHQ#CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE CorpHQ#CorpHQ# I need help to reslove the problem with TEI_ASSIGNED on the T1 Link ISDN Serial0/0/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni L2 Protocol = Q.921 0x L3 Protocol(s) = CCM MANAGER 0x0003 Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x8007 Number of L2 Discards = 0, L2 Session ID = 0 Total Allocated ISDN CCBs = 0 CorpHQ# CorpHQ#show run Building configuration... Current configuration : 3755 bytes ! ! Last configuration change at 23:30:37 PDT Fri Oct 4 2013 ! NVRAM config last updated at 23:30:39 PDT Fri Oct 4 2013 ! version 12.4 no service pad no service timestamps debug uptime no service timestamps log uptime no service password-encryption ! hostname CorpHQ ! boot-start-marker boot-end-marker ! logging message-counter syslog enable secret 5 $1$QYKS$Z.r9xOUE1ga6IkB6fi3QU0 ! no aaa new-model clock timezone PST -8 clock summer-time PDT recurring network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 ! dot11 syslog 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 service voip sip bind control source-interface Loopback0 bind media source-interface Loopback0 ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! voice-card 0 ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller T1 0/0/0 cablelength short 133 pri-group timeslots 1-3,24 service mgcp description == Voice Circuit to PSTN ! controller T1 0/0/1 cablelength short 133 channel-group 0 timeslots 1-24 description == Data Circuit to WAN ! ip tcp synwait-time 5 ! ! ! ! interface Loopback0 ip address 177.1.254.1 255.255.255.255 ! interface FastEthernet0/0 description == To SW1 no ip address duplex auto speed auto ! interface FastEthernet0/0.10 description == Server VLAN encapsulation dot1Q 10 ip address 177.1.10.1 255.255.255.0 ip pim dense-mode ip tcp adjust-mss 1300 ! interface FastEthernet0/0.11 description == Voice VLAN encapsulation dot1Q 11 ip address 177.1.11.1 255.255.255.0 ip pim dense-mode ip tcp adjust-mss 1300 ! interface FastEthernet0/0.12 description == Data VLAN encapsulation dot1Q 12 ip address 177.1.12.1 255.255.255.0 ip tcp adjust-mss 1300 ! interface FastEthernet0/0.13 description == PSTN PHONE VLAN encapsulation dot1Q 13 ip address 177.1.13.1 255.255.255.0 ip tcp adjust-mss 1300 ! interface FastEthernet0/1 description === To PSTN ip address 177.1.19.254 255.255.255.0 duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/0/1:0 description == Frame-Relay Circuit to WAN no ip address encapsulation frame-relay cdp enable no frame-relay inverse-arp frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point description == FR To BR1 ip address 177.0.101.1 255.255.255.0 ip pim dense-mode snmp trap
Re: [OSL | CCIE_Voice] TEI_ASSIGNED
You will not be able to bring up the pri until your mgcp gw registers to cucm. In my very first reply I assumed it was already registered but I guess I should have asked. The most common reason for not registering is a mismatch on hostname. The gateway will register as either hostname (if ip domain-name not set) or hostname.domain.com (if ip domain-name domain.com). I see your hostname is CorpHQ and I didn't see the ip domain-name command in yiur posted config. In this case make sure that in cucm you list the mgcp gw as CorpHQ exactly (I always type in the correct/matching case, I'm not sure if it is actually case sensitive but why take that chance?) Use show inventory on the router and make sure cucm has the correct router model and nm/wic part numbers. If still no luck (just looking at cucm), then delete the gw in cucm and recreate it. On the gateway use ccm config server [pub ip] and ccm config to tell the gateway to download the config from cucm. After it registers you can the no ccm config and make the pri fractional. For some basics, can you ping cucm pub and sub from the gateway? Can you ping from cucm cli to the gw lo0 ip? Do you have any qos? If yes remove qos temporarily to make sure that isn't interfering. If, after all this it still doesn't work, you should take a look at router debugs and cucm traces to determine if its a config issue or a network path issue. If debugs/traces don't show any traffic you could also run a packet capture on both router and cucm to see if you are getting any traffic at all between gw and cucm. On Oct 5, 2013 2:36 PM, Anthony Nwachukwu anwachu...@accesspointafrica.com wrote: Now see how far I have gone. Stil have issues with Registration with CCM and the second backup is none. ** ** ** ** CorpHQ#show ccm-manager MGCP Domain Name: CorpHQ PriorityStatus Host Primary None First BackupRegistering with CM 177.1.10.10 Second Backup None ** ** Current active Call Manager:None Backhaul/Redundant link port: 2428 Failover Interval: 30 seconds Keepalive Interval: 15 seconds Last keepalive sent:23:10:43 PDT Oct 4 2013 (elapsed time: 05:12:56) Last MGCP traffic time: 04:23:16 PDT Oct 5 2013 (elapsed time: 00:00:23) Last failover time: 03:57:15 PDT Oct 5 2013 from (0.0.0.0) Last switchback time: 03:57:06 PDT Oct 5 2013 from (177.1.10.20) Switchback mode:Immediate MGCP Fallback mode: Not Selected Last MGCP Fallback start time: None Last MGCP Fallback end time:None MGCP Download Tones:Disabled TFTP retry count to shut Ports: 2 ** ** Backhaul Link info: Link Protocol: TCP Remote Port Number: 2428 Remote IP Address: 177.1.10.10 Current Link State: OPEN Statistics: Packets recvd: 0 Recv failures: 0 Packets xmitted: 0 Xmit failures: 0 PRI Ports being backhauled: Slot 0, VIC 0, port 0 FAX mode: cisco Configuration Error History: CorpHQ# ** ** *From:* Justin Carney [mailto:justin.s.car...@gmail.com] *Sent:* 05 October 2013 18:55 *To:* Anthony Nwachukwu *Cc:* ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com) *Subject:* Re: [OSL | CCIE_Voice] TEI_ASSIGNED ** ** Did you try the steps I outlined earlier? Does show ccm indicate your mgcp gw is regisyered to cucm? If not check your hostname and domain name (if set) on gw and double check the cucm config. If so it is registered you could take it a step further and remove the pri-group from the controller (and essentially all pri and mgcp config) reboot, then put it all back. Also, if this is your home lab (or if you have access) what does the pstn side of the pri config look like? Make both sides have the same isdn switch-type and the pstn side has isdn protocol-emulate network under serial interface. A reload of both routers wouldn't hurt either (before or after all the above). Some links that may help: http://goo.gl/QA6qaC http://goo.gl/1ctXWS On Oct 5, 2013 10:03 AM, Anthony Nwachukwu anwachu...@apafrica.com wrote: CorpHQ#CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE CorpHQ#CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE CorpHQ#CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp
Re: [OSL | CCIE_Voice] translation-rule
I agree with Marty's response. I happen to be a visual learner, so if you are too then below is a your example marked up with colors to highlight the different parts of the rule. (Also, read this: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml ) voice translation-rule 1 rule 1 /^\(*12*\)3\(*45*\)$/ /6\1\2/ Set 1: *12* Default set 0: 3 (note, if you have \0 in the replace string I'm not sure if that would carry over the 3 or the full match set 12345 - it would be worth testing) Set 2: *45* router#test voice translation-rule 1 12345 Matched with rule 1 Original number: 12345 Translated number: 6*12**45* Walking through this rule left to right... 1. rule 1 /[match string]/ /[replace string]/ 2. your match string is 12345, with no digits before 1 or after 5, broken up into 2 named sets as listed above in green (set 1) and blue (set 2). 3. your replace string is 6\1\2. 4. the 6 is a literal 6 and is the first digit of the translated number. 5. next is \1 - the \ means the next character is special, so don't use it literally (ie, it's not a 1 it is instead set 1). The match string already defined set 1 as *12* by using the \( to to start the set and \) to close the set. You don't specify a number for the set - working left to right the first set is \1 second is \2 and so on. (If you don't specify any sets using \( and \) then you still have a default set 0 called as \0 in the replace string which would be used to insert the entire match string.) 6. at this point your translated number is 6 *12* (plus the remaining string). 7. next and final part of the replace string is \2 which means set 2 8. in the replace string that means put in the contents of set 2 or *45 *. 9. your translated number is 6*12**45* *Further notes, if needed:* · The use of ^ means starts with so you only match a string * starting* with 12345. o Input 12345 = MATCH, output is 61245 o Input 012345 = NO match, output is unchanged 012345 · The use of $ means ends with so you won't match any additional digits, and your string cannot contain any more digits. o Input 12345 = MATCH, output is 61245 o Input 123456 = NO match, output is 123456 The combination of using ^ and $ in this case means only match literal 12345 with nothing before or after. if you remove both ^ and $ you could match 99912345000 and get the output 99961245000. Hope this helps. If it doesn't, read the link at the top :-) On Fri, Oct 4, 2013 at 2:03 PM, Martin Sloan martinsloa...@gmail.comwrote: Hi Anthony, I'm not sure how to deep to go on the explanation but basically you have 2 capture groups in the 'match' string which are denoted by the parentheses, which have to be escaped by the backslash. These translations are based on the Unix Stream EDitor (SED) program and certain metacharaters need to be escaped to work properly, like the parentheses. They're called capture groups because whatever is included between the parentheses will be 'captured' to a buffer. You can then refer to it in the 'replace' string by referencing it's capture group number, which also has to be escaped with a backslash, like '\1'. In the *nix OS, you can create named capture groups so you can better identify the capture group and also insert new groups without having to update all others, but I don't believe this is possible in IOS. The '6' in your replace string is a literal 6. HTH Marty On Fri, Oct 4, 2013 at 1:30 PM, Anthony Nwachukwu anwachu...@apafrica.com wrote: I need with Translation -rule can someone help me explain the translation rule below. voice translation-rule 1 rule 1 /^\(12\)3\(45\)$/ /6\1\2/ · Set 1: 12 · Set 2: 45 · Ignore: 3 router#test voice translation-rule 1 12345 Matched with rule 1 Original number: 12345 Translated number: 61245 ___ 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] unity connection - transfer option for users.
What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515)? Take a look your gateways settings for Redirecting Number IE Delivery (RDNIS) for both inbound/outbound. These checkboxes are adjacent to the Display IE Delivery (which is usually turned on). To test and understand the behavior of these settings I would recommend ticking these boxes on/off and retrying your inbound/outbound calls in this (and other) scenario. As a test try setting up a call such as PSTN SA phone CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS. I haven't tested this recently and not sure if it applies in your stated scenario but try checking the box on SA gateway for the outbound RDNIS. This should allow CUC to send 3 IE out to the PSTN - the original ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be either the DN of original called IP phone or possibly the vm port/pilot's DN/EPNM) and lastly the DNIS should be 9515. For a different scenario with SB in SRST - when a call to a SB phone does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw you need to allow (check) the inbound RDNIS. In this case the IE at SA router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's DN. -Justin On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.com wrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ 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://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ 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] unity connection - transfer option for users.
I wasn't sure RDNIS would matter here but figured I would throw it out there anyway (as it applies when redirecting TO CUC). It seems the unity service parameter mentioned earlier obviates the need to use RDNIS. With the option you proposed on creating a new RP/RL just for this requirement I would just set the digit manipulation/TON on the RL to whatever you see inbound from that specific PSTN ANI to HQ - unless the question told you what the expected outbound ANI/TON should be. Another option would be to compare the original PSTN number with the destination PSTN and set to local if same NPA, LD if different NPA, or international different country codes. If it comes in unknown/unknown then send it back out that way. On Wed, Oct 2, 2013 at 12:48 PM, Martin Sloan martinsloa...@gmail.comwrote: The RDNIS shouldn't be a factor here. I just labbed this up and there is no Redirecting Number IE in the ISDN messages for this scenario. It's more of a straight dial from Unity. I think the places to be checked are: CUCM service parameter Call Routing Path Whatever Route Pattern - Route List is being used needs to have the Use Calling Party's External Phone Number Mask checked and no masking being done below, like truncating the calling party number to 7 digits if that was part of the requirement for the sites local PSTN dialing. I recommend partitioning out a new pattern that matches the number you're trying to dial and handling the digit manipulation separately from the rest of the dial plan to keep it conceptually simple, but not necessarily 'cleaner'. Kind of along the lines of keeping AAR, CFUR, SNR separate. As for the calling party TON on this, your guess is as good as mine. If the task doesn't specifically ask to set the calling party TON and it says to use any line from the PSTN phone, what do you do? Marty On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney justin.s.car...@gmail.comwrote: What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515)? Take a look your gateways settings for Redirecting Number IE Delivery (RDNIS) for both inbound/outbound. These checkboxes are adjacent to the Display IE Delivery (which is usually turned on). To test and understand the behavior of these settings I would recommend ticking these boxes on/off and retrying your inbound/outbound calls in this (and other) scenario. As a test try setting up a call such as PSTN SA phone CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS. I haven't tested this recently and not sure if it applies in your stated scenario but try checking the box on SA gateway for the outbound RDNIS. This should allow CUC to send 3 IE out to the PSTN - the original ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be either the DN of original called IP phone or possibly the vm port/pilot's DN/EPNM) and lastly the DNIS should be 9515. For a different scenario with SB in SRST - when a call to a SB phone does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw you need to allow (check) the inbound RDNIS. In this case the IE at SA router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's DN. -Justin On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.comwrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website
Re: [OSL | CCIE_Voice] QOS WAN
A few comments inline below in RED... On Tue, Oct 1, 2013 at 11:00 AM, virajith vir...@rediffmail.com wrote: Hi Justin, Thanks for your reply. After taking out and reapplying the config it appears that the HQ to SC issue for calls work. The calls between SB , HQ , SC also work. However when inbound calls come from pstn to the H323 gateway leading to the PSTN I notice that it gives a fast busy. Upon removing QOS the calls work fine. Justin: look at the response below for question 3 on LAN QoS...if I come up with something in particular I'll let you know. Take a close look at your singling path between the gateway/CUCM and the phone/CUCM in the example as both are crossing the WAN (unless this is your HQ site), while the media gateway/phone should be local (unless somehow you are using an MTP at a different site?). One-way cRTP over WAN will cause issues, but this *should* only apply to calling over WAN and not to a gateway talking to a local phone (ie, not crossing wan - unless you have a home lab where using HQ switch to power SB/SC phones across WAN). Questions: = 1) Would you recommend applying wan qos manually? is it the same procedure ? Also how is it different from the auto qos option? Justin: I always use auto qos when the question asks for MLP. It can be done manually but there are a lot of commands to build the virtual-template and I cannot do these faster than auto qos. For FRTS I sometimes use auto qos and sometimes don't (use SRDN) to be familiar with both. If the question asks for class based then I just use the SRND reference config and do it manually. 2) What would be a safe approach to take ? Justin: To avoid issues I save the running config just before and just after auto qos, then use show archive config diff flash:before flash:after to see what auto qos actually did. (I do this on the switch too.) After auto qos I edit as needed, and if it doesn't work after a few minutes of troubleshooting I reload the router and revert to the before config. After it comes back up you can either retry auto qos (faster if it works, but if it doesn't you'll lose more time to reload again) or apply it manually (because you already have the cli from the first time and your tweaks). 3) My LAN QOS has not been setup on the switch ? Do you think that this could cause an issue on the WAN? Justin: I don't think the lack of LAN QoS would affect much on the WAN, although without reviewing a specific scenario I don't want to suggest its not possible under a unique circumstance (I can't think of one at the moment). However, you need to consider the end-to-end QoS from phones/servers/gateways marking traffic and the switch and/or router re-marking/shaping/policing. For example, if your WAN QoS polices signaling at CS3 to 5% there are differences between the trusted and untrusted versions of auto-qos. The untrusted methods will build ACL/NBAR to mark your traffic, but the trusted method relies on a correct marking already. Phones and servers should mark signaling CS3, but this old version of IOS uses AF31. You will need to set your generated singling to CS3 for mgcp (mgcp ip qos dscp cs3 sig), dial-peers (under dial-peer voip: ip qos dscp cs3 sig), and sccp (sccp ip precedence 3). Router generated media usually defaults to EF. Regards, Vir From: Justin Carney justin.s.car...@gmail.com Sent: Tue, 24 Sep 2013 04:04:57 To: virajith vir...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QOS WAN For Site A - it looks like your serial sub interface 102 references DLCI 103, then suf-if 103 references DLCI 102...is that a typo or is that correct? Shut down one and verify that you can still get to the other (correct) site. For Site B - I noticed an interesting line in your R2 output on physical s0/1/0 it looks like it is set to 56K. I recently had this same issue but didn't have time to continue to troubleshoot. Whatever is causing that to show up may be part of your issue with your site B. The way I got myself in that situation was by configuring under the wrong subinterface (and creating one) and then no sub-if without removing the map-class. When I get time to lab I plan to recreate this and then try putting the wrong sub-if back, remove the map class, then delete the sub-if. I don't see anything that jumps out for your site C but will compare that to my results next time I lab (probably tomorrow) On Mon, Sep 23, 2013 at 11:46 AM, virajith vir...@rediffmail.com wrote: Hello Justin, Thanks for your reply. Please find the necessary outputs below... At R1 : = interface Serial0/0/0.102 point-to-point connected to R3 * sub-IF is .102 * ip address X..X.X.1 255.255.255.0http://www.rediffmail.com/cgi-bin/red.cgi?account_type=1red=http://255.255.255.0isImage=0BlockImage=0rediffng=0 ip ospf network point
Re: [OSL | CCIE_Voice] TEI_ASSIGNED
I'll start by assuming your config is correct. Then, the first thing I try is as follows: no mgcp mgcp bind media source lo0 or whatever interface you are using mgcp bind control source lo0 mgcp bind media source lo0 mgcp bind control source lo0 mgcp This *usually* clears it up, assuming the configuration is correct. If that doesn't work, take it one step further to reapply the L3 bind command no mgcp int s0/0/0:23 no isdn bind-l3 ccm mgcp bind media source lo0 or whatever interface you are using mgcp bind control source lo0 mgcp bind media source lo0 mgcp bind control source lo0 int s0/0/0:23 isdn bind-l3 ccm mgcp This should take care of it. (Note, I haven't had to use this more elaborate method in a while - if there's an error when trying to remove L3 bind from serial interface, you will need to shut the voice-port then the serial interface, run the commands above, then no shut the serial and then finally voice-port, lastly turning on mgcp). A few other notes - I typically use the ccm config method to build mgcp, then issue no ccm config after it is built from the CUCM download. I noticed recently in my home lab is that I was applying the command ccm-manager switchback immediate BEFORE letting CUCM build the full mgcp config - the issue here is that CUCM does not properly set the switchback method to immediate (even when set in the GUI, on these versions of sw/ios) and it reverts back to graceful switchback (as indicated using show ccm). To correct this I now apply this command (and all others extra ccm/mgcp command, such as ccm music) AFTER config download and no ccm config. For giggles I reapply the bind commands as well using the first set of commands at the top of this email. The last couple times I did this in practice I have not had any TEI_Assigned and now that this is my routine it takes very little extra time versus doing a minimal config and then troubleshooting mgcp later if needed. When in doubt, reset MGCP from the gui and from cli using no mgcp, mgcp. (The latter is required if you are NOT using ccm config or have removed it. Even when ccm config is left in for a full 23 channel PRI its a good idea to reset mgcp from cli.) Hope this helps... -Justin On Tue, Oct 1, 2013 at 11:04 AM, Anthony Nwachukwu anwachu...@apafrica.comwrote: Hi, I need help I am setting up MGCP on a 2811 router I am getting TE1-ASSIGNED. CorpHQ#debug isdn q921 debug isdn q921 is ON. CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 **ERROR**: L2IF_SendPkt: idb is NULL ISDN Se0/0/0:23 **ERROR**: process_rxdata:L2IF_SendPkt Failed CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 **ERROR**: L2IF_SendPkt: idb is NULL ISDN Se0/0/0:23 **ERROR**: process_rxdata:L2IF_SendPkt Failed CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 **ERROR**: L2IF_SendPkt: idb is NULL ISDN Se0/0/0:23 **ERROR**: process_rxdata:L2IF_SendPkt Failed CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 **ERROR**: L2IF_SendPkt: idb is NULL ISDN Se0/0/0:23 **ERROR**: process_rxdata:L2IF_SendPkt Failed CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 **ERROR**: L2IF_SendPkt: idb is NULL ISDN Se0/0/0:23 **ERROR**: process_rxdata:L2IF_SendPkt Failed CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 **ERROR**: L2IF_SendPkt: idb is NULL ISDN Se0/0/0:23 **ERROR**: process_rxdata:L2IF_SendPkt Failed CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 **ERROR**: L2IF_SendPkt: idb is NULL ISDN Se0/0/0:23 **ERROR**: process_rxdata:L2IF_SendPkt Failed CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 **ERROR**: L2IF_SendPkt: idb is NULL ISDN Se0/0/0:23 **ERROR**: process_rxdata:L2IF_SendPkt Failed CorpHQ# ISDN Se0/0/0:23 Q921: User RX - SABMEp sapi=0 tei=0 ISDN Se0/0/0:23 **ERROR**: L2IF_SendPkt: idb is NULL ISDN Se0/0/0:23 **ERROR**: process_rxdata:L2IF_SendPkt Failed CorpHQ# All possible debugging has been turned off CorpHQ#show isdn sta Global ISDN Switchtype = primary-ni ISDN Serial0/0/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x8007 Number of L2 Discards = 0, L2 Session ID = 6 --More-- ___ 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
Re: [OSL | CCIE_Voice] QOS WAN
At first glance those steps look ok. What do you see from show traffic and show frame-r pvc # on all three site routers? For odd audio issues I would look at LAN QoS (make sure you're not policing too low) and ensure you don't have one-way LFI or cRTP (which looks like you moved crtp to policy-map already). For SC phones not registering double check to ensure your frts is keeping that PVC at 2M and its not at the default 56k once frts was enabled on physical interface. Worst case you could back out the configuration, reload, and reapply the config either using auto qos or use your current config as a template and apply manually. You could also post you full cli configs for review. -Justin On Sun, Sep 22, 2013 at 10:30 AM, virajith vir...@rediffmail.com wrote: Hello All, Any update? I am still waiting for a reply. Please assist guys. From: virajith vir...@rediffmail.com Sent: Sat, 21 Sep 2013 07:18:00 To: ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com Subject: QOS WAN hi All, I am trying to connect HQ and SB with a 384 k frame relay PVC.Enable FRF.12 link fragmentation and interleave on the Frame Relay connections to fragment large data packets and interleave voice packets to minimize delays. Max delay between fragments should be set at 10 ms . Also provision RTP header compression. Configure LLQ between HQ and SB to ensure voice bearer traffic gets priority queue treatment voice signalling is guaranteed for 16kbps. Configure the priority queue accomodates up to 4 G729 calls between HQ and SB. Lastly, assume all bearer voice traffic has been marked with traffic CS3. Here are the steps I am following ... - 1) Going to serial interface connecting from HQ to SB on the HQ router and changing the BW to 384 R1(conf)#int ser0/1/0.X R1(config-subif)#bandwidth 384 2) Then apply auto qos voip trust under the interface R1(conf)#int ser0/1/0.X point-to-point #frame-relay interface-dlci 201 #auto qos voip trust 3) Then after the auto qos trust is applied . I remove no match ip dscp af31 4) add ...priority 47 and bandwidth 16 5) Then remove no frame-relay ip rtp header-compression 6) Add the following... map-class frame-relay AutoQoS-FR-Se0/1/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust 7) add compress ip header rtp 8) I then move to SB router and do the same steps ( 1-7) on the interface connecting to HQ 9) After this I create a class map on HQ router with Site C map-class frame-relay 2MBPS frame-relay cir 2048000 frame-relay bc 20480 frame-relay be 0 frame-relay mincir 2048000 interface Serial0/1/0.2 point-to-point description *** FR Connected to BR2 *** frame-relay interface-dlci 202 class 2MBPS Problem: - 1) My phones in Site C unregister and don't register back after the above configuration. Looks like QOS breaks phone registration 2) The calls between sites appear to be slightly delayed. 3) Audio on the phones seems to be watery. Questions : 1) What is wrong with the above config? 2) Is there an easier and safer way for achieving the task? 3) How can the above task be achieved without causing problems? Regards, Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ 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] H323 call failed in OWLE load 3
Take a look at the q931 to verify the outbound digits you are sending (assuming 911) and also the plan/type. In most workbooks the requirement for 911 is unknown for type and plan...if what your sending is isdn/subscriber and it don't match the question's requirement it could be the pstn rejecting the call due to that mismatch by design. If you have a translation profile on the outbound 911 dial peer make sure there are no unintential typos. In my home lab I don't change the config of my pstn router so some of the type/plan don't line up with all labs. If I have a call fail that I'm sure is going out correctly i will look at the pstn config and debugs an determine if its on the pstn side due - if so I just move on. If you're using proctor labs make sure the pstn is loaded with the right config for that lab and look at the config/debugs there as well. On Sep 3, 2013 7:53 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: all, what is the cause call using csim start 911 success, however call using phone in correct DP and CSS failed and got busy tone. - debug isdn q931 saying unllocated number (but it is there) - sh isdn stat = Frame relay established - int vlan 302 also UP and GW registered in UCM as H323 is this issue they put in OWLE or rack error. as each time when call failed , it give me error below %DSMP-3-INTERNAL: Internal Error : dsmpSession not found, -Traceback= 0x418F8AF8 0x41B1340C 0x40A8833C 0x415B7974 0x40A886F8 0x415B4564 0x40ED76F4 0x40ED1E50 K ___ 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] Multicast MoH.
Do you have the command ccm-manager music-on-hold applied? Even on h323 this command is required for pstn multicast moh On Aug 28, 2013 5:12 PM, Alex Mendoza aa.mend...@icloud.com wrote: Hi All. Is there a solution on this... GW (h323) is configured with Outbound Fast Start using IOS MTP software and is working good. - Media Termination Point Required box checked - Enable Outbound FastStart box checked with G711u-law 64K Also, I configure Multicast MoH for this site and is working good for calls from other IP Phones on the cluster. but PSTN calls trough this h323 GW is not, when I place the call on hold, PSTN caller hear unicast moh. To solve this issue, I need to remove MTP required form H323 CUCM config. I see this is an expected behavior, see the note from cisco doc. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsmoh.html *Note *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 o music on hold. To avoid this scenario, configure unicast MOH or Tone on Hold instead of multicast MOH. Is there a trick to get multicast on a PSTN call, when MTP required is active on H323 GW? Any thoughts? Alex ___ 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] srst mode all -- remove mode all
The command srst mode auto-provision all tells the router that when you learn info from a phone in srst mode (via SNAP) keep that info in running config. If you desire to keep all that info in running config you also need to keep the command in when saving the config. Althought I haven't tested it, I would expect that removing this auto provision all (or dn) would also remove the dynamically learned info from the running config but any command you manually typed (such as a hunt list) would stay. If you let srst learn the ephone dn dynamically, then you manually add a description (as in your original question) I would not expect than ephone dn or description to show in running config after removing the auto provision all. The link below discusses the various srst options and gives many cli examples of features you can deploy with cme as srst and using auto provision all. I would suggest revieiwng these config samples to help prepare for the lab. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesrst.html I have heard the same advice for srst in the lab from several people: configure srst, test it once and exit, then save and reload. Do not test srst again, trust your config if it worked the first time. The reason for this advice is that the older router code has some bugs and if you try to enter/exit srst multiple times the first will work but it may fail on subsequent tries. On Aug 22, 2013 1:28 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: hi Justion, then I after modify and before save, i also need to remove srst auto mode prov all under telephony service, right ? K *From:* Justin Carney justin.s.car...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com) ccie_voice@onlinestudylist.com *Sent:* Wednesday, August 21, 2013 8:37:11 PM *Subject:* Re: [OSL | CCIE_Voice] srst mode all Short answer - yes you need to save the config. Long answer - if you want to ensure the ephones, ephone-dns, and descriptions are all shown in the running config then you should save the running config to startup config at a point when all the settings you want (ie, description) are visible with show run. The two good ways to do this are to manually type the config before entering srst and save, or trigger srst then modify config as needed EXIT srst and save. It is typically recommended to NOT save the running config while srst is active (if using mgcp) because the mgcp-fallback process will remove automatically remove the serial interface subcommand isdn bind-l3 ccm when srst is triggered allow the default application (h323) to contol the serial interface. When cucm is reachable again this same process will restore the bind command to allow l3 backhaul of the serial port to cucm. So, if you save your config while in srst and then you (or the grading script) reload your router the bind command will be missing and the serial port layer 3 will not come up until you manually put it back. I just make it a habit to not save any configs while in srst mode, but if you are using h323 there may not be any issues - someone else may be able to comment on that or you could try it as an experiment and let the group know how it goes. On Aug 21, 2013 9:43 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: folks, when we have to use srst auto mode provision all and I need to change Desctiption of ephone-dn After come back to Normal mode. do we need to save the config or just leave it? K ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out 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] srst mode all
Short answer - yes you need to save the config. Long answer - if you want to ensure the ephones, ephone-dns, and descriptions are all shown in the running config then you should save the running config to startup config at a point when all the settings you want (ie, description) are visible with show run. The two good ways to do this are to manually type the config before entering srst and save, or trigger srst then modify config as needed EXIT srst and save. It is typically recommended to NOT save the running config while srst is active (if using mgcp) because the mgcp-fallback process will remove automatically remove the serial interface subcommand isdn bind-l3 ccm when srst is triggered allow the default application (h323) to contol the serial interface. When cucm is reachable again this same process will restore the bind command to allow l3 backhaul of the serial port to cucm. So, if you save your config while in srst and then you (or the grading script) reload your router the bind command will be missing and the serial port layer 3 will not come up until you manually put it back. I just make it a habit to not save any configs while in srst mode, but if you are using h323 there may not be any issues - someone else may be able to comment on that or you could try it as an experiment and let the group know how it goes. On Aug 21, 2013 9:43 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: folks, when we have to use srst auto mode provision all and I need to change Desctiption of ephone-dn After come back to Normal mode. do we need to save the config or just leave it? K ___ 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] Faxes are not working
Are there any other voice ports on this mgcp gw (pri) and is the gateway registered to cucm (show ccm-manager). If not, you need to fix that first, start with the commands below and allow cucm to build the rest of the cli: Ccm config server (ip address) Ccm config (If you have ip domain-name... set on router cli make sure cucm is configured with fqdn such as routerhostname.domainname.com or it won't register) Beyond gw registration, there is more config on the cli not shown on your email - do you have dial peers? (Show dial-peer voice summary). If not, you'll need something similar to this: Dial-peer voice 1 pots Application MGCPAPP Port 0/1/0 Dial-peer voice 2 pots Application MGCPAPP Port 0/1/1 The link below shows a sample config of setting up an fxs/fxo port on an ios gw. Have you already read this (or similar) and completed all steps? http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a008017787b.shtml The study list would be able to help more if you post your full config and provide more detail about what is working and what is not. More things to check and provide info to the group so we can help: Gw registered? Any other pri on gw working? Cucm conf good - gw, route pattern, etc and DNA says it will route the call outbound? Are both inbound and outbound fax calls failing or just one way? Can you call the fax on net (from an ip phone or other analog device) and it rings? What codecs and are you using modem passthru/relay? Lan only or lan and wan between 2 fax or fax and pstn?) On Aug 20, 2013 6:18 AM, Dharambir kumar varma dharambi...@gmail.com wrote: Hi My faxes are not working... where should i checkMy router is MGCP gateway... Below are config on Mgcp router.. voice-port 0/1/0 description Cheadle FAX station-id name Cheadle FAX station-id number 5867 caller-id enable ! voice-port 0/1/1 station-id name Cheadle FAX station-id number 5869 caller-id enable -- Regards, Dharambir Kumar ___ 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] Not able to register Softphone on BR2-CME
A few things to try are listed below...at first this sounds like a routing issue but if not that should be quick to rule out. (This is assuming the hard phones are in a different subnet than your pc/softphone.) -can you successfully ping from pc to a different ip on br2? -can you successfully ping from hq router to loopback, and a different ip on br2? (Use extended ping and specify a source ip/interface that is the default gw of the subnet where your pc is configured) - (from hq and br2) show ip route - (from hq and br2) show ip ospf nei - (from pc command prompt) ipconfig /all - (from pc command prompt) route print In general, there a a few quick steps when using ping from a pc to narrow down the issue, working from your pc out. 1. Ping 127.0.0.1 (loopback) 2. Ping ip assigned to nic itself 3. Ping the default gateway (if this fails check your nic settings and then the layer 2 path to router, and router itself) 4. Ping a remote host (if this fails try a different, known working, remote host. Also check your subnet mask) If any of the above fail you should have a clue where to investigate. If all the above work successfully the issue is probably not your pc. (The probably means it could still be something on your pc like Windows firewall, or an antivirus or hids sw, especially if using a corporate pc.) On Sun, Aug 18, 2013 at 11:48 AM, madhav bhardwaj ashumad...@gmail.comwrote: Hi Guys, Working on lab 3A .I am not able to register my soft phone on BR2 router but can register IP expert hardphone. Main problem lies that i can not ping CME TFTP IP address from PC that is loopback address on CME. Here is output of tracert: Tracing route to 10.10.110.3 over a maximum of 30 hops 1 285 ms 284 ms 286 ms 10.10.105.113 2 297 ms 296 ms 288 ms 10.10.100.1 3 10.10.100.1 reports: Destination host unreachable. Trace complete. Also CME ip address that is 10.10.202.1. ___ 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] Fwd: RSVP Max Sessions
My apologies for guessing about two sessions. I found through testing what Marcelo confirmed, sessions 1 will allow a single call. However, FOR THE LAB the number of software sessions per call is irrelevant - configure it way higher than you think you'll need (10, 100, 500), UNLESS the question tells you not to do that. If the question states don't configured DSPs that are not needed this does not apply to software MTP, as it doesn't use DSP resources. If you want to use a hardware MTP then yes you should be worried about DSPs (I would recommend using a separate transcoder and sticking with software MTP). I wanted to also confirm my prior statement about codec pass-through and understand when/why you should use it, so I did some further testing and research today. The short answer is that FOR THE LAB it does not appear to matter if you use codec pass-through, I got the same result for both with and without pass-through. (Please note, I did yet not test complicated scenarios such as a remote site phone calling to CCX which would use an MTP for RSVP then a transcoder to g711 to talk to CCX.) Either way, the CUCM region (g729) and location (bw unlimited, rsvp mandatory (with or without video desired)) must still be set properly. Personally, I don't use pass-through because it is one more variable if I need to troubleshoot and it does not help me in the lab. For the real world there are many compelling reasons to use codec pass-through (for example cisco tells you to) including fax/modem calls and sRTP, however those are not likely in the lab (I haven't seen them in any IPExpert workbooks). I expanded testing to see what effect codec pass through had on some other setups (beyond what we expect to see in a lab). For example (test 4 below), if CUCM is set to use G711 and IOS MTP has g729r8 and codec pass-through the call will setup using 96K and connect using 80K (sho ip rsvp res). Thus, codec passthrough effectively IGNORES the codec setting you have on the IOS MTP when the CUCM endpoints negotiate a codec. If the CUCM endpoints do not negotiate, then the ios mtp codec setting will kick in. Keep reading if you're bored or curious :-) --- Here's a show sccp with my config to look as sessions vs streams: HQ-RTR#sho sccp SCCP Admin State: UP Gateway Local Interface: Loopback0 IPv4 Address: 10.10.110.1 Port Number: 2000 IP Precedence: 3 User Masked Codec list: None Call Manager: 192.168.0.21, Port Number: 2000 Priority: N/A, Version: 5.0.1, Identifier: 2 Trustpoint: N/A Call Manager: 192.168.0.101, Port Number: 2000 Priority: N/A, Version: 5.0.1, Identifier: 1 Trustpoint: N/A MTP Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.0.101, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 1000, Reported Max OOS Streams: 0 max STREAMS 1000 (2 streams per session configured (500), which does indicate each session is one call with two streams) Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 RSVP : ENABLED dspfarm profile 2 mtp codec g729r8 rsvp maximum sessions software 500 associate application SCCP ! The output above max STREAMS 1000 while I configured 500 sessions - this does indicate each session is one end-to-end call with two streams, one for each side of the mtp. See attached (in a follow up email) detailed debugs of 5 test scenarios (rsvp bandwidth was increased to allow multiple/g711 calls) *First two test scenarios are relevant to the lab (with and without pass-through RSVP works as expected)* *1: region set g711, location set unlimited bw and rsvp mandatory, ios mtp set g729* -RESULT (as expected): rsvp uses *40k during setup, 24K when call connects* -note: show sccp conn shows codec as G729, phones also show G729 in use *2: same as test 1 but adding ios mtp codec pass-through* -RESULT (as expected): rsvp uses *40k during setup, 24K when call connects* -note: show sccp conn shows codec as pass-th, phones show G729 in use *Next two tests show codec pass-through ignores the codec set in IOS MTP* *3: region set g711, location set unlimited bw and rsvp mandatory, ios mtp * *no codec set** and pass-through on* -RESULT (as expected): rsvp uses *96k during setup, 80K when call connects* -note: show sccp conn shows codec as pass-th, phones show G711 in use *4: region set g711, location set unlimited bw and rsvp mandatory, ios mtp * *set g729** and pass-through on* -RESULT: *rsvp uses 96k during setup, 80K when call connects* (ios mtp codec setting ignored because both phones negotiate g711) -note: show sccp conn shows codec as pass-th, phones show G711 in use *Last test shows how without codec pass-through the
Re: [OSL | CCIE_Voice] RSVP Max Sessions
I believe each session is a call leg, but it doesnt really matter because the software sessions don't use any dsp resources so you don't need to be frugal. Many people use max sessions software 500 because that is the highest supported number and it doesn't cost any dsp. When verifying rsvp with show ip rsvp reservation command you will see two rows of info (one for each call leg). With g729 this command will show you 40k if you run the command while ringing or 24k if you run the command once the call is connected. Also, you should take out codec pass-through as that allows endpoints to negotiate codec and you want the mtp locked at g729r8 as listed. Hope this helps... -Justin On Aug 15, 2013 11:09 PM, Josh Petro josh.pe...@gmail.com wrote: Hi All, Can anyone explain the Maximum Sessions command to me, please? Google and Cisco have not helped me tonight.:) If I have the below config in two gateways registered to the CUCM, then I should be allowed two calls max, right? Is a Session considered a call, or just a call leg? dspfarm profile 5 mtp codec g729r8 codec pass-through rsvp maximum sessions software 2 associate application SCCP ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip rsvp bandwidth 64 ___ 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] MGCP TroubleShooting
Every (normal) teardown in MGCP begins with MDCX (and M: recvonly) and then teardown is complete with DLCX. There is a slight difference in debug order if you are debugging both Q931 and MGCP packets but in normal cases you should see MDCX then DLCX in all teardowns (except temp failure or other anomalies). IF PSTN side ends call (from GW perspective): 1. *trigger* PSTN side ends call 2. *Q931 RX disconnect *(PSTN to CUCM via GW) 3. *MGCP MDCX* (CUCM to GW) 4. MGCP 200 OK (GW to CUCM) 5. *MGCP DLCX *(CUCM to GW) 6. MGCP 250 OK (GW to CUCM) 7. *Q931 TX release* (CUCM via GW to PSTN) 8. *Q931 RX release complete* (PSTN to CUCM via GW) IF CUCM side ends call (from GW perspective): 1. *trigger* IP phone ends call (SCCP or SIP to CUCM) 2. *MGCP MDCX* (CUCM to GW) 3. MGCP 200 OK (GW to CUCM) 4. *Q931 TX disconnect*(CUCM via GW to PSTN) 5. *Q931 RX release *(PSTN to CUCM via GW) 6. *MGCP DLCX* (CUCM to GW) 7. MGCP 250 OK (GW to CUCM) 8. *Q931 TX release complete* (CUCM via GW to PSTN) To replicate these logs above you need to turn on only two debugs: debug mgcp packets debug isdn q931 In other words, in MGCP you always have MDCX/200 OK then DLCX/250 OK. For ISDN you always have DISCONNECT (side that hung up), RELEASE (side that didn't hang up), and RELEASE_COMPLETE (side that hung up. The side that hangs up dictates how the MGCP and Q931 messages are interleaved. Does this help? I know the real question you're asking is which answer is the lab looking for - I cannot answer that but it depends on the question's wording. I haven't passed yet but I believe I got an MGCP debug question correct based on my score report. If the question states show the MGCP/Q931 debug where the call *begins to teardown* I would personally use MDCX/DISCONNECT. If the question states ...where call *teardown is complete* then I would use the DLCX/RELEASE COMPLETE. If the question is not worded clearly then talk to the proctor - if you explain that you understand the FULL teardown process (similar to above) but you are not sure which to use, they will understand that you know what you are doing and not just fishing for an answer. Still, you may get the common use your best judgement response...because they cannot answer confirming questions. Worst case, if after all this you still weren't sure, you could include BOTH debugs in the notepad file and put comments in there explaining the start/end of the teardown. -Justin On Wed, Jul 17, 2013 at 2:22 PM, IE Target myfrnd...@gmail.com wrote: The message which indicates that an MGCP call is tearing down is MDCX or DLCX.?? Some guys say that it is DLCX no MDCX May be some one who got marks in this section can clarify 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] CME - GK - UCM bandwidth
The workaround is to change the service parameter for default intra-region codec to g729. You will then obviously need to update your regions for site a b c to use g711 rather than 'default' which is now g729. This bug mentioned above is where a gk send a call to cucm and it doesn't look at the region's setting (where you define a gk region and set to g729 for intra region) but instead only looks at the service parameter. On Apr 8, 2013 11:26 AM, Suresh Bhandari bring...@gmail.com wrote: Sergey, and all, I had hard-coded the G729 codec from UCM side, and when I did the same for voip dial peer pointing to RAS, it didn't show up, as it is the default. Tried with voice-class codec as well, but no luck. Will check the Bug as well. Thank Ramcharan for the bug id. On Mon, Apr 8, 2013 at 7:58 PM, Sergey Heyphets ser...@heyphets.comwrote: Alternatively, you can just set your GK region to use G729 within the region and G729 with all the other region (e.g. hardcode on region, rather than use system default). On Mon, Apr 8, 2013 at 9:56 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ 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 -- Suresh Bhandari ___ 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] CME - GK - UCM bandwidth
The brq parameter does does apply here (ie, won't fix the issue) since a brq is sent after a call is already connected and is requesting a *change* in bandwidth. The initial call setup is done with an arq that contains the initial bandwidth request. If you debug this issue end to end, you will see site 3 router send arq with bandwidth 16 (shown as 160), and gk will send acf to site 3 with be 16. The issue is from the go to cucm, I don't recall if the go arq is asking cucm for 16 or 128, but I know the acf from cucm to go is 128. No where in this call setup is a brq used, and the call is setup at 128k and uses g711. If you have time on your hands, it would certainly be a good exercise to try both service params and debug each...or just save yourself the trouble and use the intra region SVC param :-) On Apr 8, 2013 12:26 PM, Suresh Bhandari bring...@gmail.com wrote: I confirm that I was hitting the bug! Here are my results: 1. When I changed the SP BRQ Enabled to True I got the bandwidth while ringing state 128K and in connected state 16K. 2. When I changed the Intraregion codec to G729, I got the bandwidth - while ringing and connected - to be 16K. So, it really depends upon the question we face, whether to enable BRQ or set Intraregion codec. So, thank you guys. On Mon, Apr 8, 2013 at 8:46 PM, Suresh Bhandari bring...@gmail.comwrote: Sergey, and all, I had hard-coded the G729 codec from UCM side, and when I did the same for voip dial peer pointing to RAS, it didn't show up, as it is the default. Tried with voice-class codec as well, but no luck. Will check the Bug as well. Thank Ramcharan for the bug id. On Mon, Apr 8, 2013 at 7:58 PM, Sergey Heyphets ser...@heyphets.comwrote: Alternatively, you can just set your GK region to use G729 within the region and G729 with all the other region (e.g. hardcode on region, rather than use system default). On Mon, Apr 8, 2013 at 9:56 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ 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 -- Suresh Bhandari -- Suresh Bhandari ___ 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] CME - GK - UCM bandwidth
...looks like I didn't read you email correctly the first time and you already tried both methods :-) My recommendation (may or may not be the way the lab is graded) is that you should use intra region param. Reason is the grading script might not connect the call or it will only look at debugs for arq. In this case you lose points in the brq method. On the other hand, if a call is setup and the codec is check from the phones, then either method works. I can't think of a reason where you would want to have a diff bw when setting up the call than when connected, and why take the chance of losing points using brq? That said, it wouldn't hurt to use both params, as another question may require you to turn on brq. I would just recommend against *only* enabling brq to answer this question. On Apr 8, 2013 12:56 PM, Justin Carney justin.s.car...@gmail.com wrote: The brq parameter does does apply here (ie, won't fix the issue) since a brq is sent after a call is already connected and is requesting a *change* in bandwidth. The initial call setup is done with an arq that contains the initial bandwidth request. If you debug this issue end to end, you will see site 3 router send arq with bandwidth 16 (shown as 160), and gk will send acf to site 3 with be 16. The issue is from the go to cucm, I don't recall if the go arq is asking cucm for 16 or 128, but I know the acf from cucm to go is 128. No where in this call setup is a brq used, and the call is setup at 128k and uses g711. If you have time on your hands, it would certainly be a good exercise to try both service params and debug each...or just save yourself the trouble and use the intra region SVC param :-) On Apr 8, 2013 12:26 PM, Suresh Bhandari bring...@gmail.com wrote: I confirm that I was hitting the bug! Here are my results: 1. When I changed the SP BRQ Enabled to True I got the bandwidth while ringing state 128K and in connected state 16K. 2. When I changed the Intraregion codec to G729, I got the bandwidth - while ringing and connected - to be 16K. So, it really depends upon the question we face, whether to enable BRQ or set Intraregion codec. So, thank you guys. On Mon, Apr 8, 2013 at 8:46 PM, Suresh Bhandari bring...@gmail.comwrote: Sergey, and all, I had hard-coded the G729 codec from UCM side, and when I did the same for voip dial peer pointing to RAS, it didn't show up, as it is the default. Tried with voice-class codec as well, but no luck. Will check the Bug as well. Thank Ramcharan for the bug id. On Mon, Apr 8, 2013 at 7:58 PM, Sergey Heyphets ser...@heyphets.comwrote: Alternatively, you can just set your GK region to use G729 within the region and G729 with all the other region (e.g. hardcode on region, rather than use system default). On Mon, Apr 8, 2013 at 9:56 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Hi Suresh, I think you are hitting a known bug . Please go to service parameters -- call manager and change the following to G729 Intraregion Audio Codec Default: G729 Regards, Mohamed Gazzaz -- Date: Mon, 8 Apr 2013 19:02:14 +0545 From: bring...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - GK - UCM bandwidth Hello Experts! I have HQ as the GK and CME, CUCM publisher and subscriber are registered to GK in single zone. Further, GK Trunk is in separate region/DP with hard-coded G.729 codec with other regions/itself as well. When I call from HQ to CME side, and check sh gatek call it shows that the call is consuming 16K bandwidth, which is expected. The dial-peer to CME has g729r8 as codec (the default one). Even then, the same command displays that the bandwidth consumed is 128K. Any thoughts on what probably I missed? -- Suresh Bhandari ___ 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 -- Suresh Bhandari -- Suresh Bhandari ___ 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
Re: [OSL | CCIE_Voice] CUCM Call Routing
I don't know how the lab is graded, but I first answer the requirements of the question, which sometimes states to set proper plan/type and sometimes unknown, then for all other call flows that don't specify I set the proper plan/type for those. I do this for both ani and dnis. There are two reasons why I do this - first, I type all the CLI in notepad and configure all the routers at the beginning of the lab (after taking basic notes on gw type, # of channels, etc) and then I go through the gw and call routing sections and modify as needed. Second, it can make reading debugs a little easier as I am used to verifying plan/type for all calls and I only need to make a note of which calls require unknown as the exceptions. Hope this helps... -Justin On Mar 30, 2013 2:04 PM, CCIEing aboaz...@gmail.com wrote: Hi again for all, I have question regarding the exam, in the CUCM call routing section : If the question does not clearly mention that the Called Party Type and plan is required in some parts of the call routing points.. Is it better to configure the them (call type and plan) or to leave them on the default configuration ? Your input is appreciated ___ 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] MVA partial match issue
You can have the rd with 7 digits only and without the 9 for pstn access - use either application dial rules (match 7 digits, prefix 9) or a translation pattern to modify the rd to match your existing local route pattern. I'm not sure if there's an MVA bug in this version of cucm, but its pretty easy to configure it so that you always have a full match since you will likely have only one rd. This is what I do for the lab. A real world (for nanp) example of MVA partial match would be using e164 address for all rd (+1 npa-nxx-) and set partial match to 10 or 7 depending on whether all sites receive inbound ani as 10d for local calls or if any sites receives only 7d. This would also work for lab, but takes extra steps if you aren't already required to use + dialing For partial match to work, the rd must be longer than the inbound ani (ani 7d and rd +11d). You cannot use partial match with an ani longer than the rd (ani 10d and rd 7d), in this case your options would be to apply inbound transformation on the gateway to make rd ani shorter (ie match the rd) or make your rd longer and manipulate outbound dnis to make it route. On Mar 25, 2013 2:00 AM, donny f f.faraday...@gmail.com wrote: hi, I config the Service parameter for MVA , using partial match 7 digit . However when I dial the RD using 7 digit ,it never works. seem like UCM only take Full match. I heard this is bug, Any suggestion for the work around if still want to use partial match ? d On Sun, Mar 17, 2013 at 9:58 AM, michael.se...@compucom.com wrote: Greetings, I think you are doing everything right just need a few tweaks. Place a call from the PSTN line 2 to 3033300 and do debug isdn q931 on gateway. What digits do you see for the calling number. 7 or 10? If seeing 7 digits inbound change your Remote Destination Number to 525, without the 9. If you are seeing 10 digits inbound the NPA, NXX, TNTN change your remote destination number to XXX525, in other words match what you're seeing in the isdn debug for calling party and make that you're Remote Destination Number. Do NOT require the prefix of 9 on the Remote Destination Number. Also, under Remote Destination Information make sure you are putting a tick in Mobile Phone checkbox and a tick in the Enable Mobile Connect checkbox. Otherwise your configuration looks good. Hope you find this helpful. Michael Sears CCIE 38404 Date: Sun, 17 Mar 2013 18:23:01 +0530 From: sanity insanity networksanitytoinsan...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Mobile Voice Access not working since many days!! Message-ID: cag4zmyxmd5xj67pwv+_gpabedjoydyg+zbnmugtsfuj3nsx...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello All, I have been trying this config for MVA for close to 2 weeks now and it does not work . Here are the details The Issue : == I am trying to Intiate a Call from PSTN phone to site B gateway (H323) 3033300 it should ask for authentication once authenticated press 1 to make any 4 digit calls if it is from SB phone 1 . Make sure to display 4 digits number for calling number along with calling name SB Phone 1 they can use local gateway to make the call. Also 2nd line on PSTN phone should be used to dial 3033300 and you will prompted to login. Details: = My config is following 1) The dial-peers are set in the following way dial-peer voice 102 voip preference 2 destination-pattern 3300 session target ipv4:ip address of the CUCM Pub dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 5 pots service cmm incoming called-number 3300 no digit-strip 2) here is the MVA service url ! application service cmm http://ip address of the CUCM Pub:8080/ccmivr/pages/IVRMainpage.vxml ! 3) I am stripping 3033300 coming from pstn to last 4 digits using a translation-rule on the voice-port level . That is 3033300 becomes 3300 when it reaches CUCM. 4) On CUCM in the service parameters... Enable Mobile Voice access is set to True Mobile voice access number is 3300 Matching caller id with Remote Destination is Partial Match Number of digits of Caller ID Partial Match is 7 5) The Mobility softkey has been added for on hold and connected at the softkey template level and applied to the phone ( SB PH1) 6)At the User SB phone 1 I have enabled Enable Mobility and Enable Mobile Voice Access also selected the MAC address of the phone 7) Created a Remote Dest profile and selected user id of sb ph1 and the correct calling search space for the phone 8) Added a Remoted Destination number of 9525 9) Also went to device phone and selected the Owner User ID of SB Ph1 10) Cisco Unified Mobile Voice Access Service is running on both Sub and Pub on CUCM Questions : 1) I now dial from the pstn line 9525 on the pstn phone to
Re: [OSL | CCIE_Voice] Outgoing Calls via T1 failed
Your debug output has a few clues...but I can't recall offhand if channel 16 in that debug starts at 1 (meaning this is the 16th channel) or 0 (meaning this is the 17th channel). Do inbound calls from pstn work? If yes, its more likely the second option. In the first case, it would appear your issue is on the pstn side. Run show ISDN status and layer 2 should show multiple frame established and layer 3 should show ccm-manager (or similar). If however layer 2 shows tei assigned try the following: Mgcp bind media source lo0 Mgcp bind control source lo0 (Paste those commands twice) Int s0/0/0 No ISDN bind-l3 ccm ISDN bind-l3 ccm No mgcp Mgcp Show ISDN status (Ensure you see multi frame established) Also type show ccm and ensure the gw is registered to your cucm. If not, make sure that your hostname on the router matches what you have in cucm. If you have IP domain-name ipexpert.com in your config then you need to use the fqdn in cucm, such as r3.ipexpert.com. however if you don't have a domain name on the router then you should just have the routers hostname w/o domain such as r3. Now, for the other situation if channel 16 in the debug is really channel 17, that could be caused by using the ccm config command. With this, every time in cucm you reset the mgcp gw it will apply a no mgcp then mgcp and download the config from cucm to the router (and configure a FULL PRI). Ccm config command doesn't work with a fractional PRI, but you could use it to download all the commands, then no ccm config and change the controller commands to use timeslots 1-16 rather than 1-24. (Need to shut voice port, shut int s0/0/0 and no ISDN bind-l3 ccm, shut controller and remove timeslots command, then apply commands in reverse order using fractional timeslots). Not a solution, but just for reference, the default channel order for mgcp PRI is bottom up. If your issue is the latter (ccm config downloaded a full PRI config) and you were set to use ascending channels you would not have seen this issue until the 17th call came from pstn...in real lab you would lose points for having a full PRI I stead of fractional, even if calls did work. The point here is make sure you remove ccm config if you have a fractional PRI. Hope this helps... Justin On Mar 22, 2013 7:36 PM, Bill whl...@gmail.com wrote: Is your gateway registered in CUCM? Are you getting the proper output of your show commands? Show isdn status, show ccm Do you have int seri x/x/x Isdn bind-l3 ccm Did you try no MGCP MGCP? Can you post more of your config? Sent from my iPad On Mar 22, 2013, at 4:52 PM, CCIEing aboaz...@gmail.com wrote: Hi all, After configuring my HQ GW as MGCP, then configure my T1 to register with cucm , I was govern by the lack of the DSP resources, which force me to define only 16 channel out of the 23 on my pri-group under the T1 controller configuration !! here is the config : controller T1 0/0/0 pri-group timeslots 1-16 service mgcp Then I faced a problem with my outgoing calls , the calls was dropping due to the cause *Requested circuit/channel not available* My Question here, as there is 16 channel in my Pri-group are already configured, why all calls get dropped with cause of non availability of the resources, Why not to use one of the available channels (1-16) Appreciate your help here is below the output of debug q931 for one of my outgoing calls: *SDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F * *Bearer Capability i = 0x8090A2 * *Standard = CCITT * *Transfer Capability = Speech * *Transfer Mode = Circuit * *Transfer Rate = 64 kbit/s * *Channel ID i = 0xA98390 * *Exclusive, Channel 16 * *Display i = 'HQ PH1' * *Calling Party Number i = 0x2181, '7772022001' * *Plan:ISDN, Type:National * *Called Party Number i = 0x81, '911' * *Plan:ISDN, Type:Unknown* *ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x800F * *Cause i = 0x82AC1810 - Requested circuit/channel not available* ___ 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] CCIE Voice Vol1 Task 5.7
A few points to clarify, although it seems the question has already been answered... Route selection uses longest match, however the urgent priority flag allows you to route a call as soon as there is a match on that pattern, and the system will not look for a longer match. (In production use sparingly, typically just used for 911.) Partition order - this ONLY matters when you have two identical matches in two or more partitions. If your LD pattern is in a pt higher in the CSS than the block 900, the urgent priority would still force the match to the block pattern. If you had a route LD and a block LD (with the exact same string) then which ever pt is first will be used. In the event you have both a line CSS and device CSS (ie, in the real world) the pt in the line are listed before the device CSS. In this case you typically would put block pt on the line CSS and your local gw pt on the phone CSS. The exception to this is CTI route point, where device CSS is first and line CSS is second. Two good methods to troubleshoot dial plan: 1. Use dialed number analyzer (DNA) Http://pub IP/DNA 2. Dial from a phone using two methods, overlap and en bloc. Overlap is picking up handset (or speaker button) hearing dial tone and dialing one digit at a time. In this method cucm evaluates digit matches one at a time as you dial. En bloc means all at once and is when you dial the full string of digits while on hook, the you either go off hook or press the dial soft key. With this method cucm is forced to evaluate the entire digit string at once. When troubleshooting a string that should route but doesn't, go off hook and dial slowly, listening for the error tone. The position in the string where you get the error will help you find the issue. On Mar 17, 2013 8:09 AM, Bill whl...@gmail.com wrote: Ok I just tested this again and there must be something wrong with your configuration. I set this up for my HQ phones but depending on your setup you could do it for any phones In cucm create a Route Pattern of 91900, and assign it to the proper partition (maybe none or HQ) Now assign a GW (I used SLRG) Click Block this pattern [Precedence Level Exceeded] Check boxes Provide outside Dial Tone and Urgent Priority As soon as I pick up a phone and dial 91900 it stops and plays the recording. So if this is not working it could be that you do not have it in the right partition so in never gets used or you could have a replication issue. Sent from my iPad On Mar 16, 2013, at 8:51 PM, Tony Zunt tony.z...@gmail.com wrote: Evidently, I didn't read the requirement. Sorry. Would a BLOCK partition on the 900 route pattern along with a called number mask containing a phantom DN do the trick? The DN could be a cti route point forwarded to a Unity call handler which could recite the desired message. On Saturday, March 16, 2013, Tony Zunt wrote: Vignesh Associate the 900 route pattern with a partition called BLOCK. Then add BLOCK to the top of your calling search spaces. I wouldn't think it necessary to do this. Did you select the radio button for 'Do not route this pattern' on the 900 RP? Thanks On Saturday, March 16, 2013, wrote: I have already set that and tried but no luck Bill. Sent from Yahoo! Mail for iPhone Can you try making the block pattern a urgent priority pattern Sent from my iPad On Mar 16, 2013, at 1:05 PM, vignesh sethuraman sethuvign...@yahoo.co.in wrote: Hello Experts, I am working on Task 5.7 from Vol1. Question is to block the 91900?numbers. I have configured a Route pattern to block this number but this Route pattern is overridden by a another Route pattern 9.1[2-9]XX[2-9]XX which I have created for Task 5.6. I understand the longest match wins but I need to make it work as said in the question to play the error message the precedence used is not authorized for your line. Could you please let me know is there a way to make this work as expected in the question or am I missing something. Thanks, Vignesh ___ 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] DHCP WITH ONE POOL USING STATIC MAPPING.
I have used both methods and prefer the TCL shell cli as it saves a few minutes (of clicking through CUCM gui to upload the file). If you don't want to remember that CLI you can reference the origin file directly on the TFTP server (origin file tftp://[pub or sub ip address]/origin.txt) rather than copying down to the router's flash, but this may only save you a few seconds. Just another option to consider. One important note, make sure there is a SPACE in the CIDR notation between the IP and the /24 because it will not work without a space. Between the other elements I use tabs (haven't tried with spaces but that may work) and make sure you keep the header/footer and comment lines in the file. On Thu, Mar 14, 2013 at 11:01 AM, Sergey Heyphets ser...@heyphets.comwrote: Another way to upload file to router flash would be to use TCL shell, as described here -- http://blog.ioshints.info/2008/01/copy-text-files-into-router-flash.html Might save you couple of minutes. Sergey On Thu, Mar 14, 2013 at 10:12 AM, michael.se...@compucom.com wrote: Requirements: I want to configure a DHCP server on a router . The requirement is that just 1 DHCP pool is required for the phone. I am also asked to assign ip addresses of 14.10.66.13 and 14.10.66.14 for my phones. Solution: #conf t Enter configuration commands, one per line. End with CNTL/Z. ROUTER1(config)#ip dhcp database flash:origin.txt ROUTER1(config)#no service dhcp ROUTER1(config)#service dhcp ROUTER1(config)#do more origin.txt *time* Mar 14 2013 06:54 AM *version* 4 !IP address Type Hardware address Lease expiration VRF !IP address Type Hardware address Interface-name !IP address Interface-name Lease expiration Server IP address Hardware address Vrf *end* Open Notepad and modify origin.txt as below: *time* Mar 14 2013 06:55 AM *version* 4 !IP address TypeHardware address Lease expiration 142.102.66.13 /24 id 010024142EFF10 infinite 142.102.66.14 /24 id 016C504DDACC3D infinite *end* Don't forget the mask or it won't work and no dots are required in the Hardware Address. Copy the file to Publisher using tftp. Download the file to Router flash using tftp from Publisher. no ip dhcp database flash:origin.txt ip dhcp excluded-address 142.102.66.1 142.102.66.12 ip dhcp excluded-address 142.102.66.15 142.102.66.254 ip dhcp pool voice origin file flash:origin.txt option 150 ip [CUCM or CME IP Address] depends on if you're doing CUCM or CME default-router [ip address of Voice VLAN] Hope this helps Michael Sears CCIE (V) #38404 ___ 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] DHCP snooping test results
After playing with DHCP snooping for a while (i.e. pulling my hair out) , I have come up with an additional command that seems to be needed (on the router) if the scenario dictates that snooping needs to remain enabled on the switch. In my case I am using a 3560, not a 3750, and according to a lot of documentation i found the ip dhcp snooping command syntax and functionality varies a bit by switch platform. These results from my 3560 may or may not be identical to the 3750's behavior... If you have had issues of CUCM not assigning DHCP to Site A phones (or assigning to Site B but not Site A), you probably had DHCP snooping enabled on the Site A switch. After successfully breaking Site A DHCP via CUCM I setup the Site A router with local IOS DHCP, and this was broken by snooping as well. So, if CUCM DHCP doesn't work but IOS DHCP does, your issue is more likely to be with the CUCM server (possibly run utils csa disable if you've already verified the config and restarted the DHCP service.) If some sites get CUCM DHCP and others don't, or a site doesn't get any DHCP from CUCM or IOS, it's probably snooping. I would suggest using show commands to quickly see if snooping is enabled (sho run | i snoop), and if you're allowed to disable it that's the easy way - if not, practice reading through these debugs when snooping is interfering and again once you've worked around it. I found references in the solutions for OWLE labs 1 (p17) and 3 (p17) that state you can simply apply ip dhcp snoop trust on the switch ports facing the server (CUCM) and the router (a trunk). In the solutions for lab 3 there was also a test performed with an interface VLAN on the router, but when the DHCP request originates from the phone the behavior is slightly different – it seems that a switch with dhcp snooping enabled will set the giaddr to 0 in the option 82 header, and routers will typically ignore packets with the giaddr set to 0. Applying this command on the router tells it not to drop those packets with giaddr set to 0 as was done by my 3560 (I have not yet tested on a 3750 as I don't have immediate access to one): *Router# ip dhcp relay information trust-all* The following debugs helped identify these issues: *Switch3560# debug ip dhcp snooping packet* *Router# debug ip dhcp snooping packet * Attached is a text file showing relevant configs and commented debugs. I didn't want to put all that text in the email body because no one would want to read that email :-) Does anyone with a 3750 feel like testing this to see if it's relevant on that switch platform as well? Even if it's not required for a 3750 environment, I don't see how applying the command ip dhcp relay information trust-all on the router would break anything, and I'm going to add it to my routine script for Site A CLI. Hope this helps someone (besides me)... -Justin ! = 3560 switch, 12.2(46)SE Adv IP Services = ip dhcp snooping vlan 20 ip dhcp snooping-- DHCP snooping enabled, need a few commands to work around if not permitted to turn it off vlan 10 name DATA ! vlan 20 name PHONES ! vlan 30 name SERVERS interface GigabitEthernet0/3 description Phone3 switchport access vlan 10 switchport mode access switchport voice vlan 20 spanning-tree portfast ! interface GigabitEthernet0/41 description HQ-SERVER-MacBook -- CUCM running as a VM here on laptop switchport access vlan 30 switchport mode access switchport voice vlan 20 speed 100 duplex full no cdp enable spanning-tree portfast ip dhcp snooping trust -- ADD LINE HERE to trust the DHCP server interface ! interface GigabitEthernet0/47 description HQ-RTR-Fa0-0 -- trunk to router with L3 interface where ip helper-address is applied switchport access vlan 30 switchport trunk encapsulation dot1q switchport trunk native vlan 10 switchport trunk allowed vlan 10,20,30 switchport mode trunk switchport voice vlan 20 speed 100 duplex full no cdp enable spanning-tree portfast ip dhcp snooping trust -- ADD LINE HERE to trust the DHCP server interface ! ! The following debug shows ip dhcp snooping processing packet, inbound to switch (phone on Gi0/3) and floods vlan 20 (router on Gi0/47) ! note there is nothing else coming back from the router or DHCP server Switch3560#debug ip dhcp snooping packet *Mar 1 01:44:39.141: DHCPSNOOP(hlfm_set_if_input): Setting if_input to Gi0/3 for pak. Was not set *Mar 1 01:44:39.141: DHCPSNOOP(hlfm_set_if_input): Clearing if_input for pak. Was Gi0/3 *Mar 1 01:44:39.141: DHCPSNOOP(hlfm_set_if_input): Setting if_input to Gi0/3 for pak. Was not set *Mar 1 01:44:39.141: DHCP_SNOOPING: received new DHCP packet from input interface (GigabitEthernet0/3) *Mar 1 01:44:39.141: DHCP_SNOOPING: process new DHCP packet, message type: DHCPREQUEST, input interface: Gi0/3, MAC da:
Re: [OSL | CCIE_Voice] Custom Tones
I haven't tested this recently, but it may help to make the join/leave tones use different frequencies, as well as using different time intervals for the cadence. I'm not sure why you're getting these strange results (two tones on join when your cadence only shows one and no tone on leave), but there may be some strange feature (or bug) that has to do with both join and leave using the same frequency. voice class custom-cptone leave dualtone conference frequency 300 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 cadence 800 -Justin On Sun, Feb 17, 2013 at 1:56 PM, William Bell b...@ucguerrilla.com wrote: I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910 translate calling 9101 translate called 910 ! voice translation-profile 911 translate calling 9111 translate called 911 ! voice translation-profile 97 translate calling 971 translate called 97 ! voice translation-profile strip translate called 1 ! ! voice-card 0 dsp services dspfarm ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 ! controller E1
Re: [OSL | CCIE_Voice] Custom Tones
I just had another idea...you are using the *dual* tone ( ie two tones/frequencies) command, but only specified one frequency. Try adding a second number on each frequency line. voice class custom-cptone leave dualtone conference frequency 300 350 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 750 cadence 800 If this works (using two tones on the frequency command lines) then my first idea of using different values may not apply but it could be useful to troubleshoot. On Feb 17, 2013 9:45 PM, Jason Lee jas7...@gmail.com wrote: I'll give it a go tomorrow. I already reverted my pod this evening. I'll be doing another lab tomorrow, so I should be able to test this put by tomorrow afternoon. Sent from my iPad On Feb 17, 2013, at 9:14 PM, Bill whl...@gmail.com wrote: I think Justin might be on to it but it has been a while since I have done this in the lab. Sent from my iPad On Feb 17, 2013, at 3:06 PM, Justin Carney justin.s.car...@gmail.com wrote: I haven't tested this recently, but it may help to make the join/leave tones use different frequencies, as well as using different time intervals for the cadence. I'm not sure why you're getting these strange results (two tones on join when your cadence only shows one and no tone on leave), but there may be some strange feature (or bug) that has to do with both join and leave using the same frequency. voice class custom-cptone leave dualtone conference frequency 300 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 cadence 800 -Justin On Sun, Feb 17, 2013 at 1:56 PM, William Bell b...@ucguerrilla.comwrote: I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any
Re: [OSL | CCIE_Voice] [OSL | Automated Alternative Routing]
Yes, AAR is triggered on CAC reporting out of bandwidth. (Side note - the phone will display Network Congestion. Rerouting and this is a service parameter that can be customized, in case that is part of the question requirement.) You are also correct that both phones must be registered to the same CUCM cluster. I don't understand if your last sentence is a question - for some reason that call fails to call by extension - if there is a CSS/PT issue where phone A can't see the DN of phone B, AAR will not kick in. Under normal conditions phone A must be able to call phone B, then when there is no more bandwidth (per CAC) AAR will reroute via PSTN. If the phone B were in SRST mode and the WAN was down, not congested, this would instead use CFUR to reroute. I'll answer question 2 first. A common way to achieve AAR is to use a separate CSS/PT just for AAR, along with an AAR Group assigned to both lines (you can assign AAR group to phones for other reasons, but you *must* put the AAR group on the line/DN). When AAR is triggered (CAC), the called phone B's external number mask will be the new DNIS which should be in E.164 format already, and the calling phone A's AAR-CSS will be used to lookup a route for that DNIS. Simply put a \+.! route pattern in your AAR-PT that routes to the LRG, and the AAR-CSS should contain this AAR-PT. This gets the call to the gateway. If the gateway is MGCP, you may need to manipulate the plan/type to match what the PSTN expects (You may also/instead need to have a \+1.! pattern in AAR-PT in the event your MGCP router's PRI expect a 10 digit DNIS.) For H323 don't do any digit manipulation here, used the gateway to perform all manipulations. Question 1, dial peers needed. If using the strategy above, you might not need any new dial peers. For the MGCP sites there are no dial peers on the router so you are done after CUCM routes the call to the gateway in the proper format. For the H323 sites that need to route the AAR call, the DNIS will be the E.164 number when the call gets to the inbound voip dial peer. If you have an existing outbound pots dial peer that will match this E164 number there is nothing extra to do, your AAR call should be working. (make sure you have the appropriate number of digits and type/plan sent to the PSTN for both ANI and DNIS). If your existing dial peers do not match, you have a few options: 1. you could *use a translation-profile on the inbound voip dial peer *to manipulate the DNIS into something that matches an existing outbound POTS dial peer 1. for example if your DNIS is +1 408 555 1234, you would change the +1 to 91 and you would match the existing long distance outbound dial peer 2. you could *add a new outbound dial peer that will match this DNIS*(optionally putting a translation-profile on this dial peer if you need a specific plan/type) 1. for example if your DNIS is +1 408 555 1234, you can copy your existing long distance dial peer (9+11 digits) and just remove the 9 (leaving 11 digits) 3. A third option (I would recommend you do NOT use this option) would be to use number expansion to manipulate the DNIS between the inbound voip dial peer match and the outbound voip dial peer match - the reason I don't recommend this is because number expansion ALWAYS takes place between the inbound and outbound dial peers even if you don't want it to. This means if you're not careful it could break something else that was already working correctly. For question 3, TEHO - if you use the method above, your TEHO patterns will be not be visible to the AAR-CSS and there will be no conflict between TEHO patterns and the new AAR pattern. If you didn't have an isolated AAR CCS and PT, then it would be problematic if your AAR number in E164 format was matching a TEHO pattern that would send the call over the congested WAN link that trigger AAR in the first place - this obviously would not work. Hope this helps...if it doesn't I'm probably too tired to make sense and I'll edit my response tomorrow :-) -Justin On Fri, Feb 1, 2013 at 10:04 PM, ie ravindra ieravin...@gmail.com wrote: Hi Mates, As per my understanding AAR is triggering when if CAC enabled and if for some reason that call cannot be completed. Therefore we need route that call to the PSTN. The mandatory requirement is the both extensions must register to the same call manager clusted. If Caller A calling to User B. for Some reason that call fails to call by extension. AAR group grabs long number(AAR Number) from User B's Settings. and Dialls out. I have following Questions in the above. 1. when AAR is enabled if we have 3 sites, do we need to have 3 dialpeers to dialout the call. 2 What is the remomend way to make a dialplan for Multisite deployment. 3. As per my knowledge TEHO patterns should not be conflicted. If I am wrong on above statements please connect me. Thanks for Helping US, Ravi.
Re: [OSL | CCIE_Voice] [OSL | Automated Alternative Routing]
I noticed a typo in that last email (sorry, I clicked send too soon) - in the 3 options for H323 digit manipulation, I said num-exp will be between the inbound voip dial peer and outbound *VOIP *dial peer...*the outbound dial peer is POTS* in this case, not voip. (using two voip dial-peers on both inbound and outbound is CUBE, which is not relevant to AAR configuration). -Justin On Fri, Feb 1, 2013 at 11:46 PM, Justin Carney justin.s.car...@gmail.comwrote: Yes, AAR is triggered on CAC reporting out of bandwidth. (Side note - the phone will display Network Congestion. Rerouting and this is a service parameter that can be customized, in case that is part of the question requirement.) You are also correct that both phones must be registered to the same CUCM cluster. I don't understand if your last sentence is a question - for some reason that call fails to call by extension - if there is a CSS/PT issue where phone A can't see the DN of phone B, AAR will not kick in. Under normal conditions phone A must be able to call phone B, then when there is no more bandwidth (per CAC) AAR will reroute via PSTN. If the phone B were in SRST mode and the WAN was down, not congested, this would instead use CFUR to reroute. I'll answer question 2 first. A common way to achieve AAR is to use a separate CSS/PT just for AAR, along with an AAR Group assigned to both lines (you can assign AAR group to phones for other reasons, but you *must* put the AAR group on the line/DN). When AAR is triggered (CAC), the called phone B's external number mask will be the new DNIS which should be in E.164 format already, and the calling phone A's AAR-CSS will be used to lookup a route for that DNIS. Simply put a \+.! route pattern in your AAR-PT that routes to the LRG, and the AAR-CSS should contain this AAR-PT. This gets the call to the gateway. If the gateway is MGCP, you may need to manipulate the plan/type to match what the PSTN expects (You may also/instead need to have a \+1.! pattern in AAR-PT in the event your MGCP router's PRI expect a 10 digit DNIS.) For H323 don't do any digit manipulation here, used the gateway to perform all manipulations. Question 1, dial peers needed. If using the strategy above, you might not need any new dial peers. For the MGCP sites there are no dial peers on the router so you are done after CUCM routes the call to the gateway in the proper format. For the H323 sites that need to route the AAR call, the DNIS will be the E.164 number when the call gets to the inbound voip dial peer. If you have an existing outbound pots dial peer that will match this E164 number there is nothing extra to do, your AAR call should be working. (make sure you have the appropriate number of digits and type/plan sent to the PSTN for both ANI and DNIS). If your existing dial peers do not match, you have a few options: 1. you could *use a translation-profile on the inbound voip dial peer *to manipulate the DNIS into something that matches an existing outbound POTS dial peer 1. for example if your DNIS is +1 408 555 1234, you would change the +1 to 91 and you would match the existing long distance outbound dial peer 2. you could *add a new outbound dial peer that will match this DNIS*(optionally putting a translation-profile on this dial peer if you need a specific plan/type) 1. for example if your DNIS is +1 408 555 1234, you can copy your existing long distance dial peer (9+11 digits) and just remove the 9 (leaving 11 digits) 3. A third option (I would recommend you do NOT use this option) would be to use number expansion to manipulate the DNIS between the inbound voip dial peer match and the outbound voip dial peer match - the reason I don't recommend this is because number expansion ALWAYS takes place between the inbound and outbound dial peers even if you don't want it to. This means if you're not careful it could break something else that was already working correctly. For question 3, TEHO - if you use the method above, your TEHO patterns will be not be visible to the AAR-CSS and there will be no conflict between TEHO patterns and the new AAR pattern. If you didn't have an isolated AAR CCS and PT, then it would be problematic if your AAR number in E164 format was matching a TEHO pattern that would send the call over the congested WAN link that trigger AAR in the first place - this obviously would not work. Hope this helps...if it doesn't I'm probably too tired to make sense and I'll edit my response tomorrow :-) -Justin On Fri, Feb 1, 2013 at 10:04 PM, ie ravindra ieravin...@gmail.com wrote: Hi Mates, As per my understanding AAR is triggering when if CAC enabled and if for some reason that call cannot be completed. Therefore we need route that call to the PSTN. The mandatory requirement is the both extensions must register to the same call manager clusted
Re: [OSL | CCIE_Voice] [OSL | CCIE_VOICE] Location Based Call Admission Control
The two types of CAC are locations-based and RSVP-based. On CUCM they both have a configuration set on the location page, which may seem confusing at first. The key words to watch for in the scenario will be once bw is exceeded reroute call over PSTN means you need AAR, which could be either locations or RSVP. It they explicitly say use RSVP or if they say once bw is exceeded the call should proceed over WAN but get remarked down to [given DSCP] then this means you need to use RSVP. * * *When using RSVP, do not put a bandwidth on the location page in CUCM!* *Locations-based CAC setup:* -configure the location page with the bandwidth for the requested number of calls (24*Number of calls, do not add 16K for call setup) -that's it! CUCM will keep track of how many calls are going in/out of each location *RSVP-based CAC setup:* -first, you still go to the location page in CUCM, but DO NOT PUT A BANDWIDTH on the location -in the lower section of the locations page, select another location, use the reservation drop down box: - mandatory (video required) - not relevant for the lab, CUCM will try to setup audio and video and call will fail if not enough bw (then AAR will kick in if configured) - this *should* work the same as the next option since we're not using video phones, but I would suggest using the next option instead - mandatory (video desired) - use this option if you want out-of-bandwidth to reroute over PSTN (using AAR) - optional - use this option if you want the call to get remarked to a best-effort or CS1 and still go over the WAN - no reservation - this means RSVP is disabled -once the location is setup to use RSVP, you need to create an MTP for both sides of the RSVP call. For example, if using Site A to Site C, create an MTP for site A, add to MRG/MRGL and assign to site A phones, then create an MTP for Site C add to MRG/MRGL and assign to site C phones -now on to the routers, you will need to create an MTP on each router. go to the documentation site CUCM config examples and text notes ctrl+F to search for MTP and grab the IOS CLI from here. modify the sample CLI in notepad and make sure to add the rsvp command under the dspfarm profile # mtp. -finally, go to the serial sub-interface and assign the command ip rsvp bandwidth X. On this line the bw is 24*Number of calls + 16 (call setup for a single call -make sure the IOS MTP is registered in CUCM -when placing a call, use show ip rsvp reservation to watch the RSVP in progress. while ringing the output will show 40k, once connected it will show 24K. Hope this helps -Justin On Wed, Jan 30, 2013 at 5:48 AM, Suresh Bhandari bring...@gmail.com wrote: Again it depends on if you are asked to do so. On Wed, Jan 30, 2013 at 3:56 PM, ie ravindra ieravin...@gmail.com wrote: Hi All, Do we need to enable ip rsvp bandwidth command when we configure location based CAC. Thanks, Ravi. ___ 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 -- Suresh Bhandari ___ 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] [OSL | CCIE_VOICE] Location Based Call Admission Control
One more comment on CAC, but for the real world not for the lab... Locations CAC is quick and easy to setup, you don't touch the routers at all. The downside is it is really designed for a hub and spoke topology and it does not account for multiple circuits. RSVP on the other hand is path-aware, meaning you can have redundant circuits with different bandwidths. If you have a primary WAN with an rsvp bandwidth 1000 and a backup link (cable, DSL, etc) with an rsvp bandwidth 100 then the specific routers in the path of the call setup will be used to either admit the call or reroute via AAR. During normal conditions you can admit a lot of calls, but during failover where only backup circuit is active you can only admit a few calls. On Wed, Jan 30, 2013 at 8:03 AM, Justin Carney justin.s.car...@gmail.comwrote: The two types of CAC are locations-based and RSVP-based. On CUCM they both have a configuration set on the location page, which may seem confusing at first. The key words to watch for in the scenario will be once bw is exceeded reroute call over PSTN means you need AAR, which could be either locations or RSVP. It they explicitly say use RSVP or if they say once bw is exceeded the call should proceed over WAN but get remarked down to [given DSCP] then this means you need to use RSVP. * * *When using RSVP, do not put a bandwidth on the location page in CUCM!* *Locations-based CAC setup:* -configure the location page with the bandwidth for the requested number of calls (24*Number of calls, do not add 16K for call setup) -that's it! CUCM will keep track of how many calls are going in/out of each location *RSVP-based CAC setup:* -first, you still go to the location page in CUCM, but DO NOT PUT A BANDWIDTH on the location -in the lower section of the locations page, select another location, use the reservation drop down box: - mandatory (video required) - not relevant for the lab, CUCM will try to setup audio and video and call will fail if not enough bw (then AAR will kick in if configured) - this *should* work the same as the next option since we're not using video phones, but I would suggest using the next option instead - mandatory (video desired) - use this option if you want out-of-bandwidth to reroute over PSTN (using AAR) - optional - use this option if you want the call to get remarked to a best-effort or CS1 and still go over the WAN - no reservation - this means RSVP is disabled -once the location is setup to use RSVP, you need to create an MTP for both sides of the RSVP call. For example, if using Site A to Site C, create an MTP for site A, add to MRG/MRGL and assign to site A phones, then create an MTP for Site C add to MRG/MRGL and assign to site C phones -now on to the routers, you will need to create an MTP on each router. go to the documentation site CUCM config examples and text notes ctrl+F to search for MTP and grab the IOS CLI from here. modify the sample CLI in notepad and make sure to add the rsvp command under the dspfarm profile # mtp. -finally, go to the serial sub-interface and assign the command ip rsvp bandwidth X. On this line the bw is 24*Number of calls + 16 (call setup for a single call -make sure the IOS MTP is registered in CUCM -when placing a call, use show ip rsvp reservation to watch the RSVP in progress. while ringing the output will show 40k, once connected it will show 24K. Hope this helps -Justin On Wed, Jan 30, 2013 at 5:48 AM, Suresh Bhandari bring...@gmail.comwrote: Again it depends on if you are asked to do so. On Wed, Jan 30, 2013 at 3:56 PM, ie ravindra ieravin...@gmail.comwrote: Hi All, Do we need to enable ip rsvp bandwidth command when we configure location based CAC. Thanks, Ravi. ___ 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 -- Suresh Bhandari ___ 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] Workbook 1 Lab 5a
To add on to Bill's comment the inverse is also true - once you assign a gateway to a route group it will no longer be available to select as a route option inside a route pattern configuration. On Jan 28, 2013 6:35 PM, William Bell b...@ucguerrilla.com wrote: Nathan, Did you by chance assign the gateway directly to a pattern? If a gateway is assigned directly as the route option for a pattern, it is no longer available for assignment to route group. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Jan 28, 2013, at 5:42 PM, Nathan Silvers wrote: Has anyone seen when trying to create a route group the gateways do not all appear as available devices, they appear when configuring route patterns but not under route groups. -- The biggest mistake people make in life is not trying to make a living at doing what they most enjoy. - Malcolm Forbes Nathan Silvers ___ 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
