Re: [OSL | CCIE_Voice] Vol2 Lab3 Qos - HQ and BR1 - Not enough bandwidth
Try moving the ip rsvp settings to the virtual-template interface. On 3 Nov 2010, at 01:40, David A david.a...@gmail.com wrote: Hi All, Without QOS RSVP worked fine. After I enabled MLP LFI between HQ and BR1 I cannot make calls from HQ to BR1 and vice versa. I get Not Enough Bandwidth on the phones. The RSVP configuration is normal and worked before I added WAN QOS.I have reloaded the gateways. When I remove the RSVP location on the CUCM it works fine. Here are my configs HQ=== class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority 61 compress header ip rtp class AutoQoS-VoIP-Control-Trust bandwidth 16 class class-default fair-queue ! interface Serial0/0/0 no ip address encapsulation frame-relay frame-relay traffic-shaping frame-relay lmi-type ansi ip rsvp bandwidth 112 ! interface Serial0/0/0.1 point-to-point bandwidth 384 ip pim dense-mode snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/0-201 auto qos voip trust fr-atm ip rsvp bandwidth 112 ip rsvp signalling dscp 46 ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.1 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-Trust ! map-class frame-relay AutoQoS-FR-Se0/0/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 ! BR1 class-map match-any AutoQoS-VoIP-Control-UnTrust match access-group name AutoQoS-VoIP-Control class-map match-any AutoQoS-VoIP-RTP-UnTrust match protocol rtp audio match access-group name AutoQoS-VoIP-RTCP ! ! policy-map AutoQoS-Policy-UnTrust class AutoQoS-VoIP-RTP-UnTrust set dscp ef priority 61 compress header ip rtp class AutoQoS-VoIP-Control-UnTrust set dscp af31 bandwidth 16 class class-default fair-queue ! interface Serial0/0/0 no ip address encapsulation frame-relay frame-relay traffic-shaping frame-relay lmi-type ansi ip rsvp bandwidth 112 ! interface Serial0/0/0.1 point-to-point bandwidth 384 ip pim dense-mode snmp trap link-status frame-relay interface-dlci 101 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/0-101 auto qos voip fr-atm ip rsvp bandwidth 112 ip rsvp signalling dscp 46 ! ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.2 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-UnTrust ! ! ip access-list extended AutoQoS-VoIP-Control permit tcp any any eq 1720 permit tcp any any range 11000 11999 permit udp any any eq 2427 permit tcp any any eq 2428 permit tcp any any range 2000 2002 permit udp any any eq 1719 permit udp any any eq 5060 ip access-list extended AutoQoS-VoIP-RTCP permit udp any any range 16384 32767 ! ! map-class frame-relay AutoQoS-FR-Se0/0/0-101 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 ! Kindly help. Thanks, DA ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Thank you
Well done Mark, you deserved it after all the effort you put in. Thanks to you as well for your contributions to this list. Graham On 2 Nov 2010, at 23:17, Mark Holloway wrote: I want to say thank you to everyone on the OSL who has participated in any of my discussions or helped resolve issues that I encountered. I went to San Jose for my second attempt on Friday and received the news yesterday that I passed. CCIE #27384. Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
Still not sure I follow you, by the redirecting Calling Party number mask do you mean the External Phone Number Mask for the phone which is unregistered, because that doesn't work for me - I just get the For and By as the internal 4-digit number. If its some other field can you point me to where it is please Graham On 18 Oct 2010, at 11:06, sisiaji wrote: hey guys, i truly have no clue what you are talking about :))) what VM has to do with CFUR? For and By fields are representing what you have configured as redirecting Calling Party number mask (in this case redirecting ip phone) and what you have configured as a destination for such calls (Unregistered). if both are set with +... then both will be shown as +... in For/By fields... it is not a rocket science I would say so... however, for CFUR, you have to be extremely careful, as it doesn't require separate partitions/CSS to work, but if you think about it, it is the only way to fine tune it to what you want. so don't overcomplicate it, set your Unregistered destination to be +19723033001 and assuming your calling party mask is already the same, then you just need to create RP for the same + number inside separate partition which will be the only one present in a separate CSS, which in turn will need to be assigned to Unregistered Destination CSS. nothing else. when you create RP for +..., you just need to do proper digit manipulation depending on which location gateway calls is supposed to go out. so if this is national call, then you have to put inside RG/RL manipulation pre-dot (for +1.972XX), called type National, plan isdn) and don't touch calling party xformations at all as by default they are set on callmanager which means only internal 4 digits will be sent as calling numbers (that is what you see inside brackets). ok? :) On Mon, Oct 18, 2010 at 1:51 AM, Mark Holloway m...@markholloway.com wrote: I think the main thing to understand is that it should work using E164 in For/By under normal circumstances and everything else we are suggesting is a work around to a known bug with CUCM 7.0 and VMWare. On Oct 17, 2010, at 3:56 PM, Daniel Berlinski wrote: Hello guys If you want to manipulate this with CUCM the place to change the redirected number is the VM profile as indicated by Mark. Alternatively you could attach an additional rule to the translation-profile plugged inbound to the POTS call leg in the branch router in SRST mode and configure it to change the redirect-called number from to the e164 that you are after. Cheers On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway m...@markholloway.com wrote: I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and VMWare. If you go to the Device Phone and click on the Site B phones Line and specifically assign the Voicemail Profile to the Line it might work. I had success a couple of times doing this, but then after resetting my rack the last time and assigning the VM profile to the Line I still had this issue. On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote: Scenario: In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits dialing in SRST.(Wan failure) I use call forward unregistered feature. When I call from HQ Phone-1 call routed through HQ Gateway. When I call from Site-C Phone-1 call routed through the GK first and then HQ Gateway. Below is the display I am getting on my Site-B phone display. Forward HQ Phone 1 (2001) For 3001 By3001 Forward Site-C Phone 1 (4001) For 3001 By3001 My question how can I achieve below display in FOR and BY field it should be E.164 number format and than 4 digits internal ID Forward (2001) For +19723033001 (3...) By+19723033001 (3...) Forward (4001) For +19723033001 (3...) By+19723033001 (3...) Thanking you in anticipation folks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cant ping CUCM from my UNITY CONNECTION SERVER
I have seen this issue with both servers running in VMWare on the same hardware not idea why but think it's probably VMWare related On 18 Oct 2010, at 13:50, Pithog Oil pithog...@yahoo.com wrote: Quite Strange, Has anyone ran into this before, i integrated CUCM with Cisco unity connection and every thing works including MWI , but when i try to ping my CUCM from the servers in Cisco unity i get a timeout/ no response. Though the AXL admin can successfully send a test. Please i need suggestion on how to fix this, i dont think its consisitent with how things should work. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] no connectivity to CUE on a session
Is interface van 400 correctly configured and up ? On 13 Oct 2010, at 09:15, Pithog Oil pithog...@yahoo.com wrote: please i cant access my CUE, despite putting in correct configs. interface Service-Engine0/0 ip unnumbered Vlan400 service-module ip address 10.10.202.2 255.255.255.0 service-module ip default-gateway 10.10.202.1 ip route 10.10.202.2 255.255.255.255 Service-Engine0/0 BR2-RTR#service-module service-Engine 0/0 session Trying 10.10.202.1, 2194 ... % Destination unreachable; gateway or host down what do you suggest i do ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
You can also modify the redirecting number on the terminating router with a translate redirect-called in a translation profile. One thing I've noticed is that when the for number is manipulated the by number changes to unknown - anyone know why this is ? Graham On 9 Oct 2010, at 17:43, Mark Holloway wrote: By the way, I don't get why it works this way, but it does work. It's just another one of those odd things you just have to know. Vik, I know you said voicemail is the only place a Redirecting number is modified, and Marcelo mentioned there is an issue CFUR and redirecting behavior in VMWare (I experience different behavior each time I reset my rack too), so as odd as it is I think it's important to know the Voicemail profile assignment is a valid fix. On Oct 9, 2010, at 9:39 AM, Mark Holloway wrote: Ok, the secret to getting it to work every time is going to Device Phone Line and setting the voicemail profile to Default (or some voicemail profile). Even though None should use the system default voicemail profile, if you don't hard-set a voicemail profile the CFUR won't always show the external mask when the call is forwarded, but if you force a voicemail profile on the Line it will work. Thanks to both of you for your help. :) On Oct 9, 2010, at 8:58 AM, Vik Malhi wrote: Mark- can you try adding a new VM Profile for 3XXX with a MASK of the full number (the # that you want to display on the Unregistered phone). The only way to manipulate the Redirecting # in UCM is using the VM Profile. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Mark Holloway m...@markholloway.com Date: Fri, 8 Oct 2010 16:14:37 -0700 To: Mark Holloway m...@markholloway.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Call Forward Unregistered I have had it working before, but it's odd because sometimes when I reset the lab rack I can get it work and other times it does not work the way I want. I'm trying to figure out if I keep overlooking something. On Oct 8, 2010, at 4:08 PM, Mark Holloway wrote: I do not want to modify 5XXX. I want to modify 3XXX (the DN that is invoking CFUR) which is the Redirecting number. On Oct 8, 2010, at 4:02 PM, Prashant Patel wrote: Hi Mark, The easiest way is to use calling party Transformation on the outbound gateway. For example - 5002 calling 3002 out of local gateway. create a pt and assign it to a css. Assign css to the gateway calling party transformation css and uncheck use dp box. Now create a calling party transformation for 5XXX in the pt and modify the ANI to use extenal mask. This will modify the ANI from 5xxx to external mask everytime the 5xxx makes a call out of that gateway. HTH Prashant On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway m...@markholloway.com wrote: I'm trying to get my CFUR to work so it shows the External Mask in the For and By part of the call presentation but instead I am only getting it to show the 4 digit extension. For example, lets say HQ 5001 calls BR1 3001 (3001 is unregistered and has CFUR set in CUCM to dial out the PSTN because that site is in SRST mode). The presentation on the BR1 phones is Forwarded HqPh1 5001, For 3001 By 3001. Instead of 3001 I want to display the External Mask. Does anyone know the proper way to do this? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___
Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3
Currently have a similar issue with the same lab - symptoms are: MVA call connects OK and calls placed to internal numbers are fine ( except 5002 but that is the number that the mobile is linked to so may be normal - why would you call yourself) Calls placed to local/ld numbers never reach the HQ MGCP gateway Calls placed to international numbers at BR2 reach the BR2 UCME and then hang up after one ring - Cause i = 0x80AF - Resource unavailable, unspecified Time for some CUCM debugs - any other ideas ? Bits from config and debug HQ RTR voice translation-rule 100 rule 1 /^5002$/ /2123942123/ ! voice translation-profile MVA translate calling 100 application service MVA http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml dial-peer voice 5010 voip translation-profile incoming MVA service mva destination-pattern 5010 session target ipv4:10.10.210.10 incoming called-number 5010 dtmf-relay h245-alphanumeric codec g711ulaw no vad CUCM 5010 in pt-internal is matched by the MGCP gateway and points to to H.323 Gateway which points to 5010 in pt-mva which is the MVA access number FROM Br2 RTR Oct 8 08:39:17.932: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num 12123945002 Oct 8 08:39:17.932: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x0082 callID = 0x8003 switch = primary-net5 interface = User Oct 8 08:39:17.936: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8 callref = 0x0082 Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Calling Party Number i = 0x0181, '12123945002' Plan:ISDN, Type:Unknown Called Party Number i = 0x80, '5621891' Plan:Unknown, Type:Unknown BR2-RTR# Oct 8 08:39:17.972: ISDN Se0/0/0:15 Q931: RX - CALL_PROC pd = 8 callref = 0x8082 Channel ID i = 0xA98383 Exclusive, Channel 3 Oct 8 08:39:17.992: ISDN Se0/0/0:15 Q931: RX - ALERTING pd = 8 callref = 0x8082 Progress Ind i = 0x8188 - In-band info or appropriate now available Oct 8 08:39:18.084: ISDN Se0/0/0:15 Q931: TX - DISCONNECT pd = 8 callref = 0x0082 Cause i = 0x80AF - Resource unavailable, unspecified Oct 8 08:39:18.096: ISDN Se0/0/0:15 Q931: RX - RELEASE pd = 8 callref = 0x8082 Oct 8 08:39:18.100: ISDN Se0/0/0:15 Q931: TX - RELEASE_COMP pd = 8 callref = 0x0082 BR2-RTR# Regards Graham On 7 Oct 2010, at 20:00, amr thabt wrote: Hi Stutz, 1- add translation rule profile to dial-p 1997 to change the calling number to be '8884343' . 2- if still have a problem , check css of RDP and may restart Mobile Voice Service I hpoe this may help HTH AMR On Thu, Oct 7, 2010 at 9:26 PM, Stutz, Bernhard st...@pandacom.de wrote: Hi, I run into the same issue. furthermore i have to hairpin the call through a h323 gateway as all incoming calls come per mgcp to the callmanager. You have then to add a H.323 gateway to the same mgcp gateway which is possible. I got following dial--peers configured: dial-peer voice 1999 voip service cmm incoming called-number 1999 dtmf-relay h245-alphanumeric codec g711ulaw ! dial-peer voice 101 voip preference 1 destination-pattern 1997 voice-class h323 1 session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric codec g711ulaw no vad Under callmanager i have 1997 as MVA Number defined at Media Ressources-Mobile Voice Access and also at service parameter When i call the mva the call comes in via mgcp, on ccm i have a route pattern that sends 1999 back to the h.323 configured gateway, then the service gets invoked. so far so good. I have remote destination configured with 8884343 and the call comes in as following: Oct 7 21:41:58.277: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x00B4 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Calling Party Number i = 0x4180, Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0xA1, '4158881999' Plan:ISDN, Type:National Oct 7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore: Calling Number=8884343, Called Number=1999, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Oct 7 21:41:58.317:
Re: [OSL | CCIE_Voice] Single Number Reach
Mark, having done some further tests, I now have this working - the key here is that the calling number transformation pattern matches the calling number at the time the route pattern was matched. So this is likely to be 2001 as I presume that the external phone number masked is applied as a transform on the route pattern. Therefore alter your calling party transform pattern to 2XXX ( or whatever the best pattern fro HQ is) and prefix the 555. Other sites will still show the full E.164 number. Graham On 1 Oct 2010, at 18:00, Mark Holloway wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
Same here , I was beginning to think that no patterns are matched in calling number transformations - but I tested with a pattern of ! and a mask of 12345 and that works. So it would appear that there is a mismatch between \+1480.! and the calling number, which does seem odd as if you leave it alone it gets sent to the PSTN as +1480XXX. It would appear that it should match as the pattern ! with XXX works, but as Mark says this doesn't do what he requires Graham On 1 Oct 2010, at 19:23, Mark Holloway wrote: The only issue with this is you don't know if the calling party is Subscriber, National, or International, so you can't use XXX because if BR2 or BR1 calls HQ3 the From number would only show the first 7 digits. On Oct 1, 2010, at 11:21 AM, sisiaji wrote: yeah, you are right, I was referring to RP/RL transformations... i tested it and i got the same in my lab so i guess, as you already mentioned before, the way to do it is to actually put Calling Party Transform Mask to be XXX on the RL (for RG member). On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m...@markholloway.com wrote: When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry
Re: [OSL | CCIE_Voice] rsvp on redundant links
I'm pretty sure this doesn't work - the cef load balancing on the router will be based on source-destination ip address pairs so the two calls will take the same path I posted some debugs on this some months ago but I'm on road at the moment and don't have access to them Graham On 13 Sep 2010, at 16:45, Randall Saborio ill2...@gmail.com wrote: Hi, From some tests I did some time back, I figured RSVP agent does not failover to redundant links or performs any load balancing. Not sure if anyone has tested it. Will also have to contrast it when using RSVP without RSVP agent, or if CUCM will failover to other RSVP agents available through the MRGs. On Mon, Sep 13, 2010 at 8:37 AM, Stutz, Bernhard st...@pandacom.de wrote: Hi, I am trying to setup rsvp on 2 redundand links (Vol2-Lab5-5.1). I configured both links with 64k bandwidth but i don't see there load balancing happening. All calls will go via the first link and 2nd link is not been utilized. What needs to be configured to have load balancing occuring? HQ-RTR#sh ip rsvp interface interfaceallocated i/f max flow max sub max Se0/0/1:0.1 48K64K 64K 0 Se0/0/1:048K1152K1152K0 Se0/0/1:0.2 0 64K 64K 0 Se0/0/1:0.3 0 112K 112K 0 A 3nd call triggers Not enough bandwidth available. cheers, Bernhard ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IP Phone Agent service
Can you get normal login to work? if you can then the service is probably ok What urls do you have in place ? Graham On 12 Sep 2010, at 19:51, Tam Nhu tamnhu...@gmail.com wrote: Hi Experts, I am working on the UCCX and have been spent so much time to build the UCCX lab server, but I keep getting this error when trying to login via Single Button Login for agent. Cannot connect to the IP Phone Agent service. I checked all the subsystem and they are up and running; in fact, the Script and Application is working fine since I can test with the solution scripts provided in Lab 6 to Lab 10 and get the prompt and queue fine. Just that I cannot get the single button login to work. I checked again and again the configurations in UCM and UCCX, and could not find what step I am missing; probably I blind at this point. One thing I keep having the Desktop LDAP Monitor Service and Desktop Sync Service failed to start up. I redo the server, installed fresh, and still have the same problem, so I am sure that I missed something, but could not identify it. Do I need to have those two services up to be able for agent to login? I googled it but not find a useful link to fix the login issue. Thanks in advance for any helps and inputs. TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IP Phone Agent service
What happens if you point a web browser at the login URL ? You should get an XML document with the login parameters Is it a web server issue ? On 12 Sep 2010, at 20:35, Tam Nhu tamnhu...@gmail.com wrote: No, the normal login doesn't work either. Single button login URL is http://10.30.30.8:6293/ipphone/jsp/sciphonexml/IPAgentLogin.jsp Normal login URL is http://10.30.30.8:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp 10.30.30.8 is my UCCX server. Both give the same error on every phone. Thanks. TN ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unable to make GK calls from UCM to CME
Also you appear to be calling to 4001 rather then from 4001 to 852. On 11 Sep 2010, at 21:23, Ryan Schwab schwab...@shaw.ca wrote: KatGuru, In your gatekeeper main 10 debug, it states “No tech prefix”. You need a match on your tech prefix to route the call to the appropriate gateway. (Check page 524 on the CUCM SRND for GK Address resolution). In your route pattern, are you prefixing any digits to the call #? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of KatGuru Sent: September-11-10 1:27 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unable to make GK calls from UCM to CME Experts please help below is my issue GK calls from UCM to CME failed.CME to UCM works fine. I need to route the calls from HQ with the tech prefix 852. Am i missing something? GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 172.10.1.14 1720 172.10.1.14 53964 GKVOIP-GW H323-ID: gk-trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.10.1.15 1720 172.10.1.15 33266 GKVOIP-GW H323-ID: gk-trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.10.22.254 1720 172.10.22.254 62516 GKH323-GW E164-ID: 4321 H323-ID: BR2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CUCLAB-HQ(config-gk)#do sh gatek gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 1* Zone GK master gateway list: 172.10.1.15:1720 gk-trunk_2 172.10.1.14:1720 gk-trunk_1 Prefix: 852* Zone GK master gateway list: 172.10.22.254:1720 BR2 debug gatekeer main 10 (calling from HQ 4001) Sep 11 19:03:14.558: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Sep 11 19:03:14.558: ////GK/gk_rassrv_arq: arqp=0x4AF7BA9C,crv=0x4, answerCall=0 Sep 11 19:03:14.558: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/gk_dns_query: No Name servers Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/rassrv_get_addrinfo: (4001) Tech-prefix match failed. Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/rassrv_get_addrinfo: (4001) Matched zone prefix 4 and remainder 001 Sep 11 19:03:14.562: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4B3BBE78 Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/rassrv_arq_select_viazone: matched zone is GK, and z_invianamelen=0 Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4B3BBE78 Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/rassrv_arq_select_viazone: matched zone is GK, and z_outvianamelen=0 Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/rassrv_get_addrinfo: No tech prefix Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/rassrv_get_addrinfo: Alias not found Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/rassrv_get_addrinfo: (4001) unknown address and no default technology defined. Sep 11 19:03:14.562: //804A31390400/804A31390400/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed HQ gatekeeper zone local GK cisco.com 172.10.100.10 no shutdown BR2 ip address 172.10.22.254 255.255.255.0 h323-gateway voip interface h323-gateway voip id GK ipaddr 172.10.100.10 1719 h323-gateway voip h323-id BR2 h323-gateway voip tech-prefix 852 Thanks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MGCP please add to first mail on Mgcp.
Exactly, my point was that if you understood how MGCP works you would not be surprised to see this nor would you want to remove it. If you are asked in the lab to debug a non-working MGCP configuration (say for troubleshooting) then a knowledge of the MGCP protocol and messages in some details could be required, suggest you start here: http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_Guide#Troubleshooting_Cisco_IOS_Voice_Protocols This is what you would expect to see when using MGCP with an T1/E1 ISDN interface. The whole point is that MGCP backhauls the L3 signalling to CUCM. That is to say that Layer 2 signalling is handled by the router and Layer 3 signalling is handled by the CUCM. So in order to understand how to implement this fully with a Publisher/Subscriber arrangement you'll need to study things such as : How do I set up a backup CUCM? How does MGCP failover from the primary CUCM to the backup CUCM ? How is MGCP restored from the secondary to the primary CUCM ? What happens to my voice gateway if I lose connections to all the CUCM servers. How do I continue to make and receive calls? Regards Graham On 17 Aug 2010, at 11:06, Pavan wrote: It means layer 3 is being backhauled to ccm bcoz you have an isdn bind-l3 ccm-manager on the interface. Consequently router may not have any/correct information about layer 3. Sent from my phone On Aug 17, 2010, at 4:23 AM, Pithog Oil pithog...@yahoo.com wrote: Oh i think my questtion was not properly framed, i should be asking, some one to help explain what that statment means. From: Graham Hopkins ghopk...@wolf-rock.co.uk To: Pithog Oil pithog...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Mon, August 16, 2010 12:37:24 PM Subject: Re: [OSL | CCIE_Voice] MGCP please add to first mail on Mgcp. Why do you want this to stop appearing? What do you think this is saying ? Graham On 16 Aug 2010, at 06:26, Pithog Oil pithog...@yahoo.com wrote: how do i make sure this prompt stops appearing when configuring MGCP? Will this prompt affect my configurations, what is the effect of this prompt on my lab. %Q.931 is backhauled to ccm manager 0X003 on DSL1 . layer 3 output may not apply. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE #26721 - I PASSED!
Great news well done ! Regards Graham Hopkins On 17 Aug 2010, at 23:05, Matthew Berry ciscovoiceg...@gmail.com wrote: I just got my score report. I passed guys. More follow-up to come later. Right now I'm now on cloud nine. :) CCIE #26271 Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MGCP please add to first mail on Mgcp.
Why do you want this to stop appearing? What do you think this is saying ? Graham On 16 Aug 2010, at 06:26, Pithog Oil pithog...@yahoo.com wrote: how do i make sure this prompt stops appearing when configuring MGCP? Will this prompt affect my configurations, what is the effect of this prompt on my lab. %Q.931 is backhauled to ccm manager 0X003 on DSL1 . layer 3 output may not apply. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] First attempt
I don't see how this can be correct, if it is it makes the report meaningless. You could screw up a few early sections, fail on 79% and still have most of the report as 0. Of course as the score report is subject to NDA we'll never know. Still Ohamien keep working on it and you will get there. Graham On 29 Jul 2010, at 19:59, CCIE Voice GMAIL wrote: It’s also important to note, and correct me if I’m wrong, that the 0’s don’t necessarily mean you configured that section incorrectly. To my knowledge, once you lose more than 20 points, they simply stop grading your exam. So the later section may have 0’s but you configured them correctly. I feel like this is a big problem with the already vague score reports. I wish they would change this. If you are paying $1400, you deserve a full report in my opinion. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Thursday, July 29, 2010 11:25 AM To: Ohamien Uhakheme Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] First attempt I am sure you will figure out what mistakes you made which resulted in 0%. I know its very hard to find out when you are sure your solution is 100% but believe me I have been through this and you will come to know how a tiny mistake in that particular section or may be in some other section resulted in 0% for this section :) I hope you pass in 2nd attempt. Don't forget to break down your scores and analyze exactly which question you lost points. That will help you to work out on specific areas. Ash Ohamien Uhakheme wrote: Hey guys -- I've been lurking for a while, so I figured that I'd chime in. I sat for my first attempt yesterday with less than passing results. Like other people have mentioned, it is heart breaking to see 0% in areas that you are sure that you nailed completely. It's cool though, I needed to get the psychological first attempt out of the way, and I will probably schedule again for early September. IPExpert is spot on with their training material, and I definitely appreciate the effort that has gone into it. Thanks guys, Ohamien ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MWI issue with Unity Connection to UCME
try the unsolicited keyword on the mwi-server command line, without it the UCME must subscribe to the MWI service Regards Graham On 28 Jul 2010, at 11:17, Hobson Kevin wrote: Hi all, I have an issue with MWI between Unity connection and UCME. It appears that the UCME is sending back an error after the MWI notify message from the UC: Notify from UC: 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,10,TCP: destination address:port(10.10.202.1:5060)| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11,Outgoing Sip Message-- NOTIFY sip:3...@10.10.202.1 SIP/2.0| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, From: sip:3...@10.10.210.13:5060;tag=fd558ace5c314999b035bf3ba75a4893| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, To: sip:3...@10.10.202.1| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, Via: SIP/2.0/TCP 10.10.210.13:5060;branch=z9hG4bKb082ac50d8174d77a82a78f49900e05a| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, Max-Forwards: 70| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, Contact: sip:3...@10.10.210.13:5060| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, Call-ID: 23f5cf1a8c0543b0b4866f0e743ce...@10.10.202.1| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, CSeq: 300 NOTIFY| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, Event: message-summary| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, Content-Length: 23| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, Content-Type: application/simple-message-summary| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIPStack,11, Messages-Waiting: yes| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIP,11,Port=CME-1-002 prevState=SIP_IDLE newState=SIP_WAITFOR_MWINOTIFY| 09/12/2009 07:17:15.555 |8522,CME-1-002,,MiuSIP,10,CMiuSipLine::SetMWI, Wait for Notify/OK message| Response from UCME: 09/12/2009 07:17:15.576 |9310,,,MiuSIPStack,10,Received from source addr-10.10.202.1 on TCP| 09/12/2009 07:17:15.576 |9310,,,MiuSIPStack,11,Incoming Sip Message-- SIP/2.0 481 Call Leg/Transaction Does Not Exist| 09/12/2009 07:17:15.576 |9310,,,MiuSIPStack,11, Via: SIP/2.0/TCP 10.10.210.13:5060;branch=z9hG4bKb082ac50d8174d77a82a78f49900e05a| 09/12/2009 07:17:15.576 |9310,,,MiuSIPStack,11, From: sip:3...@10.10.210.13:5060;tag=fd558ace5c314999b035bf3ba75a4893| 09/12/2009 07:17:15.576 |9310,,,MiuSIPStack,11, To: sip:3...@10.10.202.1;tag=90C9BCC-7B2| 09/12/2009 07:17:15.576 |9310,,,MiuSIPStack,11, Date: Sat, 02 Mar 2002 18:10:21 GMT| 09/12/2009 07:17:15.576 |9310,,,MiuSIPStack,11, Call-ID: 23f5cf1a8c0543b0b4866f0e743ce...@10.10.202.1| 09/12/2009 07:17:15.576 |9310,,,MiuSIPStack,11, CSeq: 300 NOTIFY| 09/12/2009 07:17:15.576 |9310,,,MiuSIPStack,11, Content-Length: 0| Relevent Config fer UCME below: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip moved-temporarily sip bind control source-interface Vlan400 bind media source-interface Vlan400 registrar server voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 10 max-pool 5 timezone 23 mwi reg-e164 voicemail 3600 tftp-path flash: create profile sync 0001865842820033 ntp-server 10.10.200.2 mode unicast ! voice register dn 1 number 3005 call-forward b2bua mailbox 3005 call-forward b2bua noan 3600 timeout ! voice register pool 2 id mac 000D.BD38.7D3B type 7960 number 1 dn 1 dtmf-relay rtp-nte codec g711ulaw ! dial-peer voice 3600 voip destination-pattern 3600 session protocol sipv2 session target ipv4:10.10.210.13 dtmf-relay rtp-nte codec g711ulaw ! sip-ua mwi-server ipv4:10.10.210.13 expires 10 port 5060 transport tcp Anyone seen this? Thanks Kev On 27 July 2010 23:49, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Study partner in Vancouver area ?? (Leslie Meade) -- Message: 1 Date: Tue, 27 Jul 2010 15:49:43 -0700 From: Leslie Meade lme...@signal.ca To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Study partner in Vancouver area ?? Message-ID: 65be43a9da05cd44a3a72b458a7c0c590d6...@exch-mg.mgvfs.mcleannet Content-Type: text/plain; charset=us-ascii Is anyone else in the Vancouver area, looking for a
Re: [OSL | CCIE_Voice] CBarge in SRST mode
Thanks Daniel - decided to test with that version - everything behaves the same Regards Graham On 26 Jul 2010, at 23:55, Daniel Berlinski wrote: Guys Are you testing this particular feature in preparation for the lab exam or for your work/production requirements? I ask the question because the version of IOS all lab routers are running with is 12.4(20)T2 as per Ben Ng during the last ask the expert forum. On Tue, Jul 27, 2010 at 4:04 AM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: On my own kit 2801/2811 12.4.24Txx ( will check exact version when I get home). Yes I can only get it to work with auto provision none if I use privacy off on the ephone, it then appears to take the phone template that refers to the remote in use soft keys but no privacy button appears on the phones. I tend to agree that this must be IOS related as everyone gets slightly different results. Just wanted to explore all the options in case a lab question asked not to configure the ephones and was also thinking about the comment on the IP Expert blog - from Ben Ng I think - saying that there are bugs and we ought to know the workarounds Graham On 26 Jul 2010, at 16:44, Mark Holloway m...@markholloway.com wrote: Graham, Are you configuring this in your own lab or using Proctor Labs? I am using my own lab (2800's, 12.4.24T3, 7965 phones) and I couldn't get cBarge to work in SRST with auto provision none. Others using Proctor Labs said they could get it to work. Perhaps it's a difference between IOS versions and/or phone types. I literally tried everything. On Jul 26, 2010, at 6:59 AM, Graham Hopkins wrote: Been following the thread on this and have concerns about the ephone-template not appearing to work. The only but I can find that relates to this is CSCsx15347 which refers to a G.729 codec in the ephone -template not being used until after a reboot. The only way I can get this to work without specifying privacy off under the ephone is to run with srst mode auto-provision all and then save the config and reboot - the ephone-template then works privacy button as well . Config below. Anyone have any further thoughts on how to do this without using auto-provision all. Anyone found a way to do it with auto provision none and the ephone template - no manual configuration of the ephone? telephony-service sdspfarm units 4 sdspfarm tag 1 br1-conf no privacy conference hardware srst mode auto-provision all srst ephone template 1 srst dn line-mode octo max-ephones 4 max-dn 8 ip source-address 10.10.201.1 port 2000 system message CCIE SRST Fallback voicemail 912123945600 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jul 21 2010 11:48:33 ephone-template 1 privacy off privacy-button softkeys remote-in-use Newcall CBarge ephone 1 mac-address 0026.CB3D.2888 ephone-template 1 button 1:1 2:2 3:3 ! ! ! ephone 2 mac-address 0021.D8B8.EDDF ephone-template 1 button 1:4 2:3 ! Regards Graham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] QoS question about uplink port
As you say trust is only inbound - after all you trust what you are getting not what you are sending. With regards to uplinks then it depends - if its a trunk it will have CoS and DSCP in the tagged frames as CoS is part of the 802.1p header. If its an access link then it will have DSCP only. Regards Graham On 26 Jul 2010, at 14:23, Matthew Berry wrote: Guys - When configuring QoS on an uplink port, how do I determine whether to trust CoS or DSCP markings? I always thought that you would trust CoS markings on access ports with IP phones on the other end since the phone will mark packets as CoS3 (signaling) or CoS 5 (media). The access ports connected to servers would be configured to trust DSCP since CUCM marks according to DSCP. My understanding is that the mls qos trust cos or mls qos trust dscp applies only for inbound packets. Ideas? Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CBarge in SRST mode
Been following the thread on this and have concerns about the ephone-template not appearing to work. The only but I can find that relates to this is CSCsx15347 which refers to a G.729 codec in the ephone -template not being used until after a reboot. The only way I can get this to work without specifying privacy off under the ephone is to run with srst mode auto-provision all and then save the config and reboot - the ephone-template then works privacy button as well . Config below. Anyone have any further thoughts on how to do this without using auto-provision all. Anyone found a way to do it with auto provision none and the ephone template - no manual configuration of the ephone? telephony-service sdspfarm units 4 sdspfarm tag 1 br1-conf no privacy conference hardware srst mode auto-provision all srst ephone template 1 srst dn line-mode octo max-ephones 4 max-dn 8 ip source-address 10.10.201.1 port 2000 system message CCIE SRST Fallback voicemail 912123945600 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jul 21 2010 11:48:33 ephone-template 1 privacy off privacy-button softkeys remote-in-use Newcall CBarge ephone 1 mac-address 0026.CB3D.2888 ephone-template 1 button 1:1 2:2 3:3 ! ! ! ephone 2 mac-address 0021.D8B8.EDDF ephone-template 1 button 1:4 2:3 ! Regards Graham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge in SRST mode
On my own kit 2801/2811 12.4.24Txx ( will check exact version when I get home). Yes I can only get it to work with auto provision none if I use privacy off on the ephone, it then appears to take the phone template that refers to the remote in use soft keys but no privacy button appears on the phones. I tend to agree that this must be IOS related as everyone gets slightly different results. Just wanted to explore all the options in case a lab question asked not to configure the ephones and was also thinking about the comment on the IP Expert blog - from Ben Ng I think - saying that there are bugs and we ought to know the workarounds Graham On 26 Jul 2010, at 16:44, Mark Holloway m...@markholloway.com wrote: Graham, Are you configuring this in your own lab or using Proctor Labs? I am using my own lab (2800's, 12.4.24T3, 7965 phones) and I couldn't get cBarge to work in SRST with auto provision none. Others using Proctor Labs said they could get it to work. Perhaps it's a difference between IOS versions and/or phone types. I literally tried everything. On Jul 26, 2010, at 6:59 AM, Graham Hopkins wrote: Been following the thread on this and have concerns about the ephone-template not appearing to work. The only but I can find that relates to this is CSCsx15347 which refers to a G.729 codec in the ephone -template not being used until after a reboot. The only way I can get this to work without specifying privacy off under the ephone is to run with srst mode auto-provision all and then save the config and reboot - the ephone-template then works privacy button as well . Config below. Anyone have any further thoughts on how to do this without using auto-provision all. Anyone found a way to do it with auto provision none and the ephone template - no manual configuration of the ephone? telephony-service sdspfarm units 4 sdspfarm tag 1 br1-conf no privacy conference hardware srst mode auto-provision all srst ephone template 1 srst dn line-mode octo max-ephones 4 max-dn 8 ip source-address 10.10.201.1 port 2000 system message CCIE SRST Fallback voicemail 912123945600 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jul 21 2010 11:48:33 ephone-template 1 privacy off privacy-button softkeys remote-in-use Newcall CBarge ephone 1 mac-address 0026.CB3D.2888 ephone-template 1 button 1:1 2:2 3:3 ! ! ! ephone 2 mac-address 0021.D8B8.EDDF ephone-template 1 button 1:4 2:3 ! Regards Graham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Question about recert timelines?
June 2011 they are cumulative but there is a waiting period between recerts to stop you going crazy and getting 10 years in one month still always pays to get it done early Regards Graham Hopkins On 26 Jul 2010, at 17:55, john D jkd1...@gmail.com wrote: Hello All, I have a quick question related to recert. My current cert ends June2011. So I have 1 year to recertify. If for example I recertify before that time (lets say for example Dec2010), will the next 2 year kick in from Dec2010 or After June2011..? Thanks in advance! John ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] bad problem in configuring isdn pri
Do you have access to the device you are calling ? Are you sure it is setup and ready to receive the call ? Call it from a handset - you should hear it ring even if the call cannot complete as it's ISDN end to end The problem may be with the provider but it could also be with the terminating device Graham On 23 Jul 2010, at 06:37, Akbar Ali ccie...@gmail.com wrote: Dear Bo , i am pri only for backup of mpls link , but first i am trying to test my link from router itself. but getting error that says no route to destination . provider is not ready to accept that this is their problem. Regards Akbar On Thu, Jul 22, 2010 at 4:55 PM, Bo Gao bga...@gmail.com wrote: Do you need to set isdn incoming-voice voice on D channel? On Thu, Jul 22, 2010 at 2:49 AM, Akbar Ali ccie...@gmail.com wrote: Dear all , I am getting unexpected error while configuring isdn pri E1 , that i am not able to understand as provider says everything is right from their side what to do please help me. following is my configuration and errors also ... i tried self test on pri... controller E1 0/1/0 pri-group timeslots 1-31 description +BSNL PRI+ interface Serial0/1/0:15 no ip address encapsulation ppp dialer rotary-group 1 dialer-group 1 isdn switch-type primary-net5 no peer default ip address ppp authentication chap interface Dialer1 ip address 10.130.253.254 255.255.255.0 encapsulation ppp no ip mroute-cache dialer in-band dialer idle-timeout 9 dialer map ip 10.130.253.252 name FIS_HCBLROUTER broadcast dialer load-threshold 1 either dialer-group 1 HO-Rtr#sh log Syslog logging: enabled (0 messages dropped, 105 messages rate-limited, 0 flushes, 0 overruns, xml disabled, filtering disabled) No Active Message Discriminator. No Inactive Message Discriminator. Console logging: level debugging, 55360 messages logged, xml disabled, filtering disabled Monitor logging: level debugging, 0 messages logged, xml disabled, filtering disabled Buffer logging: level debugging, 55463 messages logged, xml disabled, filtering disabled Logging Exception size (4096 bytes) Count and timestamp logging messages: disabled Persistent logging: disabled Trap logging: level informational, 741 message lines logged Log Buffer (4096 bytes): *Jul 22 09:05:01.697: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:01.701: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:11.697: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:11.697: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:21.697: ISDN Se0/1/0:15 Q921: User TX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:21.701: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:21.701: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:21.705: ISDN Se0/1/0:15 Q921: User RX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:31.697: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:31.697: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENT: UserIdle: callid 0x80B9 received IS DN_CALL (0x0) *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: UserIdle: Call to 2320635 at 64 Kb /s *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: isdn_get_guid: Cannot allocate a G UID (5) *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENT: process_pri_call: call id 0x80B9, n umber 2320635, Guid 0026F91065D9, speed 64, call type DATA, redial No, CSM call No, pdata No *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: process_pri_call: No name in GTD *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: fill_cid_table_voice: Don't know c alling number for redial. *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: fill_cid_table_voice: Created entr y call_id 0x80B9, speed 64, remote 2320635, calling *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: Packet to CC Data *Jul 22 09:05:41.433: 4D000180B91604030800101804000300 *Jul 22 09:05:41.433: FF700900013233323036333504030800 *Jul 22 09:05:41.433: 101803000300 *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: calltrkr_setup_received: isdn_info =1732177424l, call_id=0x80B9 ORIGINATE *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: calltrkr_setup_received: calltrack er disabled *Jul 22 09:05:41.433: ISDN Se0/1/0:15 Q931: Sending SETUP callref = 0x0126 call ID = 0x80B9 switch = primary-net5 interface = User *Jul 22 09:05:41.433: ISDN Se0/1/0:15 Q921: User TX - INFO sapi=0 tei=0, ns=16 nr=16 *Jul 22 09:05:41.433: ISDN Se0/1/0:15 Q931: SETUP pd = 8 callref = 0x0126 Bearer Capability i = 0x8890 Standard = CCITT Transfer Capability = Unrestricted Digital Transfer Mode = Circuit Transfer Rate = 64 kbit/s
Re: [OSL | CCIE_Voice] bad problem in configuring isdn pri
So if I get this correct you are trying to get the router to call itself using the test call command and the service provider isn't routing the call back to you. If you call this number from a handset do you see the call reaching the router when debugging Q.931 ? Regards Graham On 23 Jul 2010, at 11:04, Akbar Ali wrote: Dear Graham , I am calling remote device , i am trying to call on the same router where pri is terminated. with command isdn test call interface serial0/1/0:15 but getting the error no route to destination while debugging isdn standard and q321 . I tried google that says This cause indicates that the called party cannot be reached because the network through which the call has been routed does not serve the destination desired. This cause is supported on a network dependent basis. Regards On Fri, Jul 23, 2010 at 12:52 PM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: Do you have access to the device you are calling ? Are you sure it is setup and ready to receive the call ? Call it from a handset - you should hear it ring even if the call cannot complete as it's ISDN end to end The problem may be with the provider but it could also be with the terminating device Graham On 23 Jul 2010, at 06:37, Akbar Ali ccie...@gmail.com wrote: Dear Bo , i am pri only for backup of mpls link , but first i am trying to test my link from router itself. but getting error that says no route to destination . provider is not ready to accept that this is their problem. Regards Akbar On Thu, Jul 22, 2010 at 4:55 PM, Bo Gao bga...@gmail.com wrote: Do you need to set isdn incoming-voice voice on D channel? On Thu, Jul 22, 2010 at 2:49 AM, Akbar Ali ccie...@gmail.com wrote: Dear all , I am getting unexpected error while configuring isdn pri E1 , that i am not able to understand as provider says everything is right from their side what to do please help me. following is my configuration and errors also ... i tried self test on pri... controller E1 0/1/0 pri-group timeslots 1-31 description +BSNL PRI+ interface Serial0/1/0:15 no ip address encapsulation ppp dialer rotary-group 1 dialer-group 1 isdn switch-type primary-net5 no peer default ip address ppp authentication chap interface Dialer1 ip address 10.130.253.254 255.255.255.0 encapsulation ppp no ip mroute-cache dialer in-band dialer idle-timeout 9 dialer map ip 10.130.253.252 name FIS_HCBLROUTER broadcast dialer load-threshold 1 either dialer-group 1 HO-Rtr#sh log Syslog logging: enabled (0 messages dropped, 105 messages rate-limited, 0 flushes, 0 overruns, xml disabled, filtering disabled) No Active Message Discriminator. No Inactive Message Discriminator. Console logging: level debugging, 55360 messages logged, xml disabled, filtering disabled Monitor logging: level debugging, 0 messages logged, xml disabled, filtering disabled Buffer logging: level debugging, 55463 messages logged, xml disabled, filtering disabled Logging Exception size (4096 bytes) Count and timestamp logging messages: disabled Persistent logging: disabled Trap logging: level informational, 741 message lines logged Log Buffer (4096 bytes): *Jul 22 09:05:01.697: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:01.701: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:11.697: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:11.697: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:21.697: ISDN Se0/1/0:15 Q921: User TX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:21.701: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:21.701: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:21.705: ISDN Se0/1/0:15 Q921: User RX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:31.697: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:31.697: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENT: UserIdle: callid 0x80B9 received IS DN_CALL (0x0) *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: UserIdle: Call to 2320635 at 64 Kb /s *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: isdn_get_guid: Cannot allocate a G UID (5) *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENT: process_pri_call: call id 0x80B9, n umber 2320635, Guid 0026F91065D9, speed 64, call type DATA, redial No, CSM call No, pdata No *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: process_pri_call: No name in GTD *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: fill_cid_table_voice: Don't know c alling number for redial. *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: fill_cid_table_voice: Created entr y call_id 0x80B9, speed 64, remote 2320635
Re: [OSL | CCIE_Voice] VMWare server
I run two machines of the spec you mention with ubuntu. No problems with CUCM UC UCCX CUPS and a client XP machine using vmware server 2 except restore from snapshot is slow if you do too many at once and MVA tends to go very sluggish when servers are heavily loaded. Just generic hardware but I do spread the load over the two servers I do run dynamips but not usually at the same time Graham On 20 Jul 2010, at 19:56, Akash akashapa...@yahoo.com wrote: Thanks Pavan for sharing your experience. Were you using dynamics as well on the server? Do you know good deal for suffient hardware requirements? Akash Patel Presales Consultant On Jul 20, 2010, at 2:36 PM, Pavan pav.c...@gmail.com wrote: I tried installing ccm 7 on vmware server 2 on top of ubuntu 10 (64 bit) couple of times and could never get install to complete successfully. On the other hand, I have used esxi and vmware workstation without any problems Sent from my phone On Jul 20, 2010, at 12:06 PM, akash patel akashapa...@yahoo.com wrote: I am planning to install CUCM Pub/Sub, UCCX, Unity Connection and Presence server on VMWare Server 2 on top of Ubantu. The reason to choose VMWare Ser 2 instead of ESXi because I was told that it works better with dynamics in order to simulate voice routers including FR and PSTN simulation. the server config I am looking in to is Intel Quad Processor 8 gig RAM two- 250G hard=drive, one for Pub and UCCX and other one for other servers Does any one has any suggestion, specifically to find out cheaper server with this or recommended hardware requirements? appreciate all feedback. thank you, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue
Yes similar to what I was doing. Also tried doing the same with a SIP gateway, which is a real pain as the SIP trunk from CUCM doesn't pass the type and plan. Also does anyone know if there is a SIP equivalent of no supplementary-service h225-notify cid-update - or any other way of preventing the 9 appearing on the phone display. Regards Graham On 13 Jul 2010, at 03:53, Mark Holloway wrote: Ok, so this is how set my H.323 gateway to operate. For example, a single POTS dial peer to handle Local calls (7 digit called, 7 digit calling number) for normal operation with UCM and when the router is in SRST mode. dial-peer voice 4 voip description Calls from UCM add 9 translation-profile incoming ADD9 incoming called-number . voice translation-profile ADD9 translate called 50 voice translation-rule 50 rule 1 /\(.*\)/ /9\1/ dial-peer voice 920 pots description LOCAL translation-profile outgoing LOCAL destination-pattern 9[2-9]..$ port 0/0/0:23 voice translation-profile LOCAL translate calling 11 translate called 10 voice translation-rule 10 rule 1 // // type unknown subscriber plan unknown isdn ! voice translation-rule 11 rule 1 /\(^2...$\)/ /222\1/ On Jul 9, 2010, at 12:22 PM, Graham Hopkins wrote: With the two sets of dial-peers you do need to take care that overlapping patterns don't cause problems in SRST for example I hit issues with [2-9].. and 91[2-9]..[2-9].. I decided to go with the translation pattern to put the 9 back on to the digits sent by CUCM, but this 9 will still show up on the phone unless you use voice service voip no supplementary-service h225-notify cid-update Regards Graham Hopkins On 9 Jul 2010, at 19:21, Mark Holloway wrote: Sounds like you have the PSTN to CUCM part working ok. This is what I have been doing. On the H323 router create the following dial-peer dial-peer voice 10 pots destination-pattern [2-9]..$ port 0/0/0:23 On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls originated by BR1 phones and strip the predot. This way you can assign the call type as Subscriber within the Route Pattern and if local calls are supposed to send a 7 digit calling number you can set the calling party transformation mask within the Route Pattern to XXX. You could have a second dial-peer on your H323 router for SRST dial-peer voice 910 pots destination-pattern 9[2-9]..$ port 0/0/0:23 translation-profile outgoing LOCAL There are really two different ways to handle H323 gateway dial-peers. You can strip the 9 in CUCM then add it back on the H323 gateway through a translation-profile and only have one set of dial-peers. Or, build your dial-peers for local, LD, international, and 911 without the 9, copy/paste in notepad and put a 9 in front of the dial-peer number and the destination-pattern then paste it into your router. You will have two sets of dial-peers for SRST and normal operation. On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote: HI Team, I have an issue with this question. Question === when pstn number 414363 call phones at site b they should display 7 digits on the phone display. For example when pstn calling ph 1 or ph 2 at branch B it should display 363 on the screen. My solution = I have added +1 in Device pool of Branch B to make it globalised when the call comes in the H323 Branch B router. I have created \+1414.363 calling party transformation mask. I have created \+1414.363 route pattern with Branch B as the gateway. (branch b is the H323 gateway). So on the Route pattern i have just done predot and in the branch b route list i have done NANP-Predot and prefix 9. I have done vice versa as well but things doesnt work. IN the branch B router i have a dial-peer for the local calls. dial-peer voice 1 pots destination-pattern 9[2-9].. port 0/0/0:23 translation-profile outgoing local translation-rule 1 rule 1 /^8.../ /363\0/ translation-rule 2 rule 1 // // type any sub plan any isdn translation-profile lcoal translate called 2 translate calling 1 Note: If i make a dial-peer without 9 i.e (...) Then the display is perfect. but i dont feel this would be the solution. because in srst this would be an issue. Issue = The issue is when PSTN phone 414363 calls Brach B ph1 or ph2 the caller id is 363 and in the missed call its globalized number +1414363 as per the question. But when i do redial using missed calls from Branch B ph1 or ph2 the calling number on the ip phones is displayed as 9363 (9 is the secondary dial tone) and the call goes through. Evrything works fine except for the display on ph1 or ph2, there is 9. How do i get rid of it 9. I hope i have made my point very clear of what issue i
Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue
Well the answer on the SIP gateway is to rewrite the SIP message using voice service voip sip sip-profiles 1 ! voice class sip-profiles 1 response 183 sip-header Remote-Party-ID modify (.*):9(.*) \1:\2 Thanks to Mark Holloway's blog for pointing me in the right direction. Jul 13 17:31:59.297: //-1//SIP/Info/sip_profiles_application_modify_remove_header: Header before modification : Remote-Party-ID: sip:95621...@10.10.110.3;party=called;screen=no;privacy=off Jul 13 17:31:59.297: //-1//SIP/Info/sip_profiles_application_modify_remove_header: Header after modification : Remote-Party-ID: sip:5621...@10.10.110.3;party=called;screen=no;privacy=off Sent: SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 10.10.210.11:5060;branch=z9hG4bK80124aba121 From: Br2 Ph1 sip:5623...@10.10.210.11;tag=bdc70633-cf9d-4ffb-8d2d-b6a883aec812-49066039 To: sip:5621...@10.10.110.3;tag=26568D0-1811 Date: Tue, 13 Jul 2010 17:31:59 GMT Call-ID: 890d9700-c3c1a30f-ad7-bd20...@10.10.210.11 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: sip:5621...@10.10.110.3;party=called;screen=no;privacy=off Contact: sip:5621...@10.10.110.3:5060;transport=tcp Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 6425 6036 IN IP4 10.10.110.3 s=SIP Call c=IN IP4 10.10.110.3 t=0 0 m=audio 19552 RTP/AVP 18 19 c=IN IP4 10.10.110.3 On 13 Jul 2010, at 08:13, Graham Hopkins wrote: Yes similar to what I was doing. Also tried doing the same with a SIP gateway, which is a real pain as the SIP trunk from CUCM doesn't pass the type and plan. Also does anyone know if there is a SIP equivalent of no supplementary-service h225-notify cid-update - or any other way of preventing the 9 appearing on the phone display. Regards Graham On 13 Jul 2010, at 03:53, Mark Holloway wrote: Ok, so this is how set my H.323 gateway to operate. For example, a single POTS dial peer to handle Local calls (7 digit called, 7 digit calling number) for normal operation with UCM and when the router is in SRST mode. dial-peer voice 4 voip description Calls from UCM add 9 translation-profile incoming ADD9 incoming called-number . voice translation-profile ADD9 translate called 50 voice translation-rule 50 rule 1 /\(.*\)/ /9\1/ dial-peer voice 920 pots description LOCAL translation-profile outgoing LOCAL destination-pattern 9[2-9]..$ port 0/0/0:23 voice translation-profile LOCAL translate calling 11 translate called 10 voice translation-rule 10 rule 1 // // type unknown subscriber plan unknown isdn ! voice translation-rule 11 rule 1 /\(^2...$\)/ /222\1/ On Jul 9, 2010, at 12:22 PM, Graham Hopkins wrote: With the two sets of dial-peers you do need to take care that overlapping patterns don't cause problems in SRST for example I hit issues with [2-9].. and 91[2-9]..[2-9].. I decided to go with the translation pattern to put the 9 back on to the digits sent by CUCM, but this 9 will still show up on the phone unless you use voice service voip no supplementary-service h225-notify cid-update Regards Graham Hopkins On 9 Jul 2010, at 19:21, Mark Holloway wrote: Sounds like you have the PSTN to CUCM part working ok. This is what I have been doing. On the H323 router create the following dial-peer dial-peer voice 10 pots destination-pattern [2-9]..$ port 0/0/0:23 On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls originated by BR1 phones and strip the predot. This way you can assign the call type as Subscriber within the Route Pattern and if local calls are supposed to send a 7 digit calling number you can set the calling party transformation mask within the Route Pattern to XXX. You could have a second dial-peer on your H323 router for SRST dial-peer voice 910 pots destination-pattern 9[2-9]..$ port 0/0/0:23 translation-profile outgoing LOCAL There are really two different ways to handle H323 gateway dial-peers. You can strip the 9 in CUCM then add it back on the H323 gateway through a translation-profile and only have one set of dial-peers. Or, build your dial-peers for local, LD, international, and 911 without the 9, copy/paste in notepad and put a 9 in front of the dial-peer number and the destination-pattern then paste it into your router. You will have two sets of dial-peers for SRST and normal operation. On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote: HI Team, I have an issue with this question. Question === when pstn number 414363 call phones at site b they should display 7 digits
Re: [OSL | CCIE_Voice] Vol2 Lab8 Call Forward to VM
I presume you have redirecting version header delivery outbound set on the SIP Trunk? On 13 Jul 2010, at 19:53, Kevin Damisch kevin.dami...@vitalsite.com wrote: This is usually a no-brainer. Working on the VM section of Vol2 Lab8. Whenever a forward busy/no answer call to 5002 goes to VM, it plays the opening greeting instead of going to the 5002 mailbox. RTMT doesn’t show any info about 5002. Caller is 5001, called is 5600, reason is Direct, and Redir/Last Redir are empty. These are SIP phones and using the SIP trunk to Unity Connection and not sure what is different about doing this compared to the old school VM port wizard method. I can access the mailbox on 5002, then choose the option to send a VM to itself, lights up MWI, I can check it, and MWI goes off. That part is good, it’s just the busy/no answer doesn’t work properly. I’ve never seen this behavior in production either. Any thoughts? This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE not stating PSTN Calling Party Number
Add the line voicemail callerid not sure where it is in the GUI - must check Graham On 12 Jul 2010, at 06:42, Mark Holloway m...@markholloway.com wrote: I'm not quite sure what's causing this issue, but when any PSTN number calls Br2Ph1 or Br2Ph2 I can see the Calling party information fine in the ISDN setup and on the display of the phones, but if I let it go to voicemail and then check messages from the phones after MWI lights up, CUE always says An unknown caller left you a message. I'm not sure why CUE isn't stating the Calling Party number? Any ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue
With the two sets of dial-peers you do need to take care that overlapping patterns don't cause problems in SRST for example I hit issues with [2-9].. and 91[2-9]..[2-9].. I decided to go with the translation pattern to put the 9 back on to the digits sent by CUCM, but this 9 will still show up on the phone unless you use voice service voip no supplementary-service h225-notify cid-update Regards Graham Hopkins On 9 Jul 2010, at 19:21, Mark Holloway wrote: Sounds like you have the PSTN to CUCM part working ok. This is what I have been doing. On the H323 router create the following dial-peer dial-peer voice 10 pots destination-pattern [2-9]..$ port 0/0/0:23 On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls originated by BR1 phones and strip the predot. This way you can assign the call type as Subscriber within the Route Pattern and if local calls are supposed to send a 7 digit calling number you can set the calling party transformation mask within the Route Pattern to XXX. You could have a second dial-peer on your H323 router for SRST dial-peer voice 910 pots destination-pattern 9[2-9]..$ port 0/0/0:23 translation-profile outgoing LOCAL There are really two different ways to handle H323 gateway dial-peers. You can strip the 9 in CUCM then add it back on the H323 gateway through a translation-profile and only have one set of dial-peers. Or, build your dial-peers for local, LD, international, and 911 without the 9, copy/paste in notepad and put a 9 in front of the dial-peer number and the destination-pattern then paste it into your router. You will have two sets of dial-peers for SRST and normal operation. On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote: HI Team, I have an issue with this question. Question === when pstn number 414363 call phones at site b they should display 7 digits on the phone display. For example when pstn calling ph 1 or ph 2 at branch B it should display 363 on the screen. My solution = I have added +1 in Device pool of Branch B to make it globalised when the call comes in the H323 Branch B router. I have created \+1414.363 calling party transformation mask. I have created \+1414.363 route pattern with Branch B as the gateway. (branch b is the H323 gateway). So on the Route pattern i have just done predot and in the branch b route list i have done NANP-Predot and prefix 9. I have done vice versa as well but things doesnt work. IN the branch B router i have a dial-peer for the local calls. dial-peer voice 1 pots destination-pattern 9[2-9].. port 0/0/0:23 translation-profile outgoing local translation-rule 1 rule 1 /^8.../ /363\0/ translation-rule 2 rule 1 // // type any sub plan any isdn translation-profile lcoal translate called 2 translate calling 1 Note: If i make a dial-peer without 9 i.e (...) Then the display is perfect. but i dont feel this would be the solution. because in srst this would be an issue. Issue = The issue is when PSTN phone 414363 calls Brach B ph1 or ph2 the caller id is 363 and in the missed call its globalized number +1414363 as per the question. But when i do redial using missed calls from Branch B ph1 or ph2 the calling number on the ip phones is displayed as 9363 (9 is the secondary dial tone) and the call goes through. Evrything works fine except for the display on ph1 or ph2, there is 9. How do i get rid of it 9. I hope i have made my point very clear of what issue i am facing. The question state the display on the phone should be only 363 and not 9363. Regards, JF ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] NTP
Default stratum is 8 so a simple ntp master will work Graham On 8 Jul 2010, at 23:59, Mark Holloway m...@markholloway.com wrote: Yikes, I meant ntp master stratum X not ntp server stratum X On Jul 8, 2010, at 3:57 PM, Mark Holloway wrote: If a router (for example, HQ) is configured with the ntp server x.x.x.x command to sync time from another source, but I want another device (such as PUB) to get its time from the HQ router, do I also need to configure the HQ router with ntp server stratum X or can UCM simply get the time sync from HQ without the stratum command? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phones on CUCM
Have you changed anything on the router? I have seen this when adding a sip trunk and binding the sip traffic to a different address than the source address used in voice register global. Regards Graham Hopkins On 7 Jul 2010, at 08:18, Duncan Hamilton-Walker wrote: Hi All, So for some very strange reason ... My SIP phones that registered to CUCM (ver 7.1.2.2-2) Have now decided to continually register and unregister themself every 10-20 seconds I have the standard SIP profile applied, this has not happen before.. Any ideas ? Thanks Duncan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phones on CUCM
Here's a lesson - read and reread the question - CUCM not CUCME, sorry. Regards Graham Hopkins On 7 Jul 2010, at 08:53, Graham Hopkins wrote: Have you changed anything on the router? I have seen this when adding a sip trunk and binding the sip traffic to a different address than the source address used in voice register global. Regards Graham Hopkins On 7 Jul 2010, at 08:18, Duncan Hamilton-Walker wrote: Hi All, So for some very strange reason ... My SIP phones that registered to CUCM (ver 7.1.2.2-2) Have now decided to continually register and unregister themself every 10-20 seconds I have the standard SIP profile applied, this has not happen before.. Any ideas ? Thanks Duncan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] B-ACD and G.729 Calls
I have an issue with codecs and a B-ACD Script, the router has a transcoder which works for calls into a CUE Autoattendant, problem appears to be with matching the correct dial-peers for calls to B-ACD. Requriement is call from WAN using G.729 talks to B-ACD - can it be done ? Setup application service app-b-acd-aa param voice-mail 3600 paramspace english index 1 param max-time-call-retry 700 param service-name app-b-acd param number-of-hunt-grps 2 paramspace english language en param handoff-string app-b-acd-aa param dial-by-extension-option 3 param max-time-vm-retry 2 paramspace english location flash: param aa-pilot 3502 param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 ! service app-b-acd param queue-len 15 param aa-hunt1 3001 param number-of-hunt-grps 2 param aa-hunt2 3002 param queue-manager-debugs 1 ! Dial-Peers for calls to B-ACD AA dial-peer voice 3503 pots service app-b-acd-aa incoming called-number 3502 port 0/0/0:15 dial-peer voice 3502 voip service app-b-acd-aa destination-pattern 3502 incoming called-number 3502 session target ipv4:10.10.110.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! PSTN and Local Calls work fine,remote calls are answered but then silence - codec issue ? Looking at dial-peer matching for calls across the WAN to the CUE AA these two are used dial-peer voice 5100 voip destination-pattern [15]... session target ras dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3500 voip description AA destination-pattern 3500 session protocol sipv2 session target ipv4:10.10.202.2 dtmf-relay sip-notify rtp-nte codec g711ulaw no vad that invokes the transcoder so tried to split the B-ACD dial-peers inbound/outbound into two thus dial-peer voice 3502 voip service app-b-acd-aa destination-pattern 3502 session target ipv4:10.10.110.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad dial-peer voice 3505 voip incoming called-number 3502 dtmf-relay sip-notify rtp-nte but I don't get transcoding I get Jul 7 10:08:56.845: %CALL_CONTROL-6-CALL_LOOP: The incoming call has a global identfier already present in the list of currently handled calls. It is being refused for both local and remote calls, guess the issue here is how to route the call in-out via the loopback and invoke the transcoder Graham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Layer 2 Overhead Size on Frame Relay
This question keeps cropping up so I thought I'd share my findings on this: QoS SRND MLP - 13 bytes FR - 4 bytes FRF.12 - 8 bytes UC SRND MLP - 10 bytes FR - 4 bytes FRF.12 - can't find it. Looking at the standards RFC 1990 for MLP Long Sequence Number Format 10 bytes Short Sequence Number Format 8 bytes So together with the UC SRND I assume Cisco use the Long Sequence Number Format and would use the 10 bytes figure FRF.12 Seems to have options for example on Cisco the End-to-End Fragmentation and Switched PVC Fragmentation formats are different: However in Cisco Press - Cisco Frame Relay Solutions Guide - I find (Figure 16.3) 2 bytes FR Header 2 bytes UI and NLPID ( Network Layer Protocol Identifier) 2 bytes Fragmentation Header 2 bytes FCS Total 8 bytes which matches the QoS SRND Normal FR is a 2 byte header and 2 byte FCS giving 4 bytes as in both SRNDs So summary - MPLP = 10 bytes, FRF.12= 8 bytes FR = 4 bytes Any other options welcome. Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ipphone does not looking for List.xml
It looks for the List.xml file when you press settingsuser preferencesbackground images not at registration time - have you tested that ? you should then see something like Looking for Desktops/320x212x12/List.xml (will vary with phone model), you can then set the tftp server alias to point to your file. Regards Graham Hopkins On 6 Jul 2010, at 10:44, jeremy co wrote: Hi, I'm trying to customize the background on my ipphone and downloaded List.xml and image and thumbnail to flash. Problem I faced from very begining is debug tftp does not show that ipphone looking for List XML at all. I reset the phone ,but same result. Jul 6 09:39:21.171: TFTP: Server request for port 49178, socket_id 0x66B007B4 for process 186 Jul 6 09:39:21.171: TFTP: read request from host 142.4.30.1(49178) via FastEthernet0/0.300 Jul 6 09:39:21.171: TFTP: Looking for CTLSEP0017E066C0CD.tlv Jul 6 09:39:21.171: TFTP: Sending error 1 No such file Jul 6 09:39:21.299: TFTP: Server request for port 49179, socket_id 0x66B007B4 for process 186 Jul 6 09:39:21.299: TFTP: read request from host 142.4.30.1(49179) via FastEthernet0/0.300 Jul 6 09:39:21.299: TFTP: Looking for SEP0017E066C0CD.cnf.xml Jul 6 09:39:21.307: TFTP: Opened flash:/its/vrf1/XMLDefault7961.cnf.xml, fd 0, size 1099 for process 186 C2801(config-ephone)# Jul 6 09:39:21.307: TFTP: Sending block 1 (retry 0), socket_id 0x66B007B4 Jul 6 09:39:21.307: TFTP: Received ACK for block 1, socket_id 0x66B007B4 Jul 6 09:39:21.307: TFTP: Sending block 2 (retry 0), socket_id 0x66B007B4 Jul 6 09:39:21.311: TFTP: Received ACK for block 2, socket_id 0x66B007B4 Jul 6 09:39:21.311: TFTP: Sending block 3 (retry 0), socket_id 0x66B007B4 Jul 6 09:39:21.311: TFTP: Received ACK for block 3, socket_id 0x66B007B4 Jul 6 09:39:21.311: TFTP: Finished flash:/its/vrf1/XMLDefault7961.cnf.xml, time 00:00:00 for process 186 C2801(config-ephone)# Jul 6 09:39:22.723: TFTP: Server request for port 49180, socket_id 0x66B007B4 for process 186 Jul 6 09:39:22.723: TFTP: read request from host 142.4.30.1(49180) via FastEthernet0/0.300 Jul 6 09:39:22.723: TFTP: Looking for English_United_States/mk-sccp.jar Jul 6 09:39:22.723: TFTP: Sending error 1 No such file Jul 6 09:39:22.867: TFTP: Server request for port 49181, socket_id 0x66B007B4 for process 186 Jul 6 09:39:22.867: TFTP: read request from host 142.4.30.1(49181) via FastEthernet0/0.300 Jul 6 09:39:22.867: TFTP: Looking for United_States/g3-tones.xml Jul 6 09:39:22.867: TFTP: Sending error 1 No such file C2801(config-ephone)# Jul 6 09:39:23.407: %IPPHONE-6-REG_ALARM: 22: Name=SEP0017E066C0CD Load= SCCP41.8-3-3S Last=Reset-Reset Jul 6 09:39:23.439: %IPPHONE-6-REGISTER: ephone-5:SEP0017E066C0CD IP:142.4.30.1 Socket:3 DeviceType:Phone has registered. Cheers, Jeremy ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ipphone does not looking for List.xml
Good point - of course UCME tells you what its looking for, but for UCM see the phone manuals (an often overlooked resource I think) Products Section Voice and Unified CommsIP TelephonyIP PhonesCisco Unified IP Phone 7900 SeriesMaintain and Operate Guides Cisco Unified IP Phone 7965G and 7945G Administration Guide for Cisco Unified Communications Manager 7.0 (SCCP and SIP)Customizing the Cisco Unified IP Phone the Desktop Folder is listed here - List.xml File Format Requirements The List.xml file defines an XML object that contains a list of background images. The List.xml file is stored in the following subdirectory on the TFTP server: /Desktops/320x212x16 Regards Graham On 6 Jul 2010, at 13:23, Mouhammad Nasser wrote: Hi Graham, I have a question here about the different folder values, is there a reference that contain all values for different models, can we access such a reference in the exam, or we have to memorize values of the 7965 series? Best regards, Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HQ BR2 - CUE Transcoding
You'll need to do it at BR2 - if you do it at HQ/BR1 it will be G.711 across the WAN. Graham On 6 Jul 2010, at 17:41, Mark Holloway wrote: If calls should complete using G.729 from HQ/BR1 to CUE on BR2 which is G.711u, can the transcoding be configured on the BR2 router locally or does it need to happen via the originating party's transcoding resources in UCM? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HQ BR2 - CUE Transcoding
Mark is that the full telephony-service below or an extract? You'll need max-dn, max-ephone and a source address to fire up sccp fully I can see the dspfarm profile has registered so just taking a guess really. However did this myself this afternoon - had some dtmf-relay issues but transcoder was ok - post the whole config if you like. Graham On 6 Jul 2010, at 18:43, Mark Holloway m...@markholloway.com wrote: Thanks, everyone. I configured the Transcoder locally on BR2. Now my issue is when I call from HQ to BR2 CUE, the call is answered by CUE but I do not hear the CUE attendant. The HQ phone shows RTP Sender packets incrementing but my Rcvr packets is not incrementing. Local BR2 phones work fine, so I know CUE is up and running. Has anyone experienced one-way audio with CUE before while Transcoding? r3-br2#show sccp Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, 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 r3-br2#show sdspfarm units mtp-3 Device:CME-XCODE TCP socket:[7] REGISTERED in SCCP ver 17/10 actual_stream:8 max_stream 8 IP:192.168.1.254 31790 MTP Dixieland keepalive 19 Supported codec: G711Ulaw G711Alaw G729a G729ab r3-br2# show run | sec teleph telephony-service sdspfarm units 5 sdspfarm transcode sessions 6 sdspfarm tag 2 CME-XCODE r3-br2#show dspfarm profile 2 Dspfarm Profile Configuration Profile ID = 2, Service = TRANSCODING, Resource ID = 2 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 4 Number of Resource Available : 4 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 On Jul 6, 2010, at 10:23 AM, Graham Hopkins wrote: You'll need to do it at BR2 - if you do it at HQ/BR1 it will be G.711 across the WAN. Graham On 6 Jul 2010, at 17:41, Mark Holloway wrote: If calls should complete using G.729 from HQ/BR1 to CUE on BR2 which is G.711u, can the transcoding be configured on the BR2 router locally or does it need to happen via the originating party's transcoding resources in UCM? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HQ BR2 - CUE Transcoding
I think Miron maybe onto something here - the two calls legs imply the transcoder is working ( show sccp conn will prove that ) but possibly the CUE cannot route outside of VLAN 500 which maybe causing issues. Check that you have the default gateway set Graham On 6 Jul 2010, at 20:34, Miron Kobelski findko...@gmail.com wrote: have you tried pinging: 192.168.50.29 with source 192.168.1.254? I can't check it right now, but there is a command which showe number of RTP packets sent/received by a router (show voip rtp connectiond detail maybe). On Tue, Jul 6, 2010 at 8:47 PM, Mark Holloway m...@markholloway.com wrote: Man, I'm stuck. :( Here is the BR2 configuration. voice-card 0 dspfarm dsp services dspfarm sccp local Vlan500 sccp ccm 192.168.1.254 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 2 bind interface Vlan500 associate ccm 1 priority 1 associate profile 2 register CME-XCODE ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729r8 codec g729br8 codec g729abr8 codec g729ar8 maximum sessions 4 associate application SCCP dial-peer voice 4000 voip description CUE destination-pattern 4000 session protocol sipv2 session target ipv4:192.168.1.253 dtmf-relay sip-notify codec g711ulaw no vad telephony-service sdspfarm units 5 sdspfarm transcode sessions 4 sdspfarm tag 2 CME-XCODE conference hardware no auto-reg-ephone authentication credential administrator cisco max-ephones 2 max-dn 10 ip source-address 192.168.1.254 port 2000 url services http://192.168.1.253/voiceview/common/login.do url authentication http://192.168.1.254/CCMCIP/authenticate.asp time-format 24 voicemail 4000 max-conferences 2 gain -6 call-forward pattern .T web admin system name administrator password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T When HQ calls CUE I'll get the following output on the BR2 router even though the HQ phone (192.168.50.29) doesn't increment Rcvr Packets. Call from HQ to Br2Ph1 or Br2Ph2 work fine (of course, no transcoding required). r3-br2#show voip rtp connections VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 17 18 1653227762 192.168.1.254 192.168.50.29 2 18 17 1844016904 192.168.1.254 192.168.1.253 Found 2 active RTP connections ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE and CUCM intergration issue
Have you given the jtapi user control of the cti ports and the rp ? Regards Graham Hopkins On 5 Jul 2010, at 10:11, Hobson Kevin wrote: Hi all, I am having real issues with this. The CUE just refuses to register to the UCM. Below is what i have done so far to try and get this working: If i do a show ccn status ccm manager i get the below: cue# sh ccn status ccm-manager JTAPI Subsystem is not registered with any Call Manager I have configured the following on UCM: ctiport - 3601 ctiport - 3602 rp-aa - 3100 rp-vm - 3600 jtapi application user - enabled standard cti enabled role. CUE config below: cue# sh run Generating configuration: clock timezone Europe/Madrid hostname cue ip domain-name ipexpert.com line console exit system language preferred en_GB ntp server 10.10.200.2 prefer software download server url ftp://127.0.0.1/ftp; credentials hidden 6u/dKTN/hsEuSAEfw40XlF2eFHnZfyUTSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP privilege vm-imap create privilege ViewPrivateList create privilege ManagePrompts create privilege broadcast create privilege ManagePublicList create privilege ViewHistoricalReports create privilege ViewRealTimeReports create privilege manage-passwords create privilege local-broadcast create privilege manage-users create groupname Broadcasters create username administrator create privilege vm-imap description Privilege to manage personal voicemail via IMAP client privilege ViewPrivateList description Privilege to view private list privilege ManagePrompts description Privilege to create, modify, or delete system prompts privilege broadcast description Privilege to send local or remote broadcast messages privilege ManagePublicList description Privilege to manage public lists privilege ViewHistoricalReports description Privilege to view historical reports privilege ViewRealTimeReports description Privilege to view realtime reports privilege manage-passwords description Privilege to reset user passwords privilege local-broadcast description Privilege to send local broadcast messages privilege manage-users description Privilege to create, modify, and delete users and groups privilege vm-imap operation voicemail.imap.user privilege ViewPrivateList operation voicemail.lists.private.view privilege ManagePrompts operation prompt.modify privilege ManagePrompts operation system.debug privilege broadcast operation broadcast.local privilege broadcast operation broadcast.remote privilege broadcast operation system.debug privilege ManagePublicList operation voicemail.lists.public privilege ManagePublicList operation system.debug privilege ViewHistoricalReports operation report.historical.view privilege ViewRealTimeReports operation report.realtime privilege manage-passwords operation user.password privilege manage-passwords operation user.pin privilege manage-passwords operation system.debug privilege local-broadcast operation broadcast.local privilege local-broadcast operation system.debug privilege manage-users operation user.password privilege manage-users operation group.configuration privilege manage-users operation user.pin privilege manage-users operation user.mailbox privilege manage-users operation user.configuration privilege manage-users operation user.remote privilege manage-users operation system.debug privilege manage-users operation user.notification groupname Administrators member administrator groupname Broadcasters privilege broadcast restriction msg-notification create restriction msg-notification min-digits 1 restriction msg-notification max-digits 30 restriction msg-notification dial-string preference 1 pattern * allowed backup server url ftp://127.0.0.1/ftp; credentials hidden EWlTygcMhYmjazXhE/VNXHCkplVV4KjescbDaLa4fl4WLSPFvv1rWUnfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP calendar biz-schedule systemschedule open day 1 from 00:00 to 24:00 open day 2 from 00:00 to 24:00 open day 3 from 00:00 to 24:00 open day 4 from 00:00 to 24:00 open day 5 from 00:00 to 24:00 open day 6 from 00:00 to 24:00 open day 7 from 00:00 to 24:00 end schedule ccn application autoattendant aa description autoattendant enabled maxsessions 6 script aa.aef parameter busClosedPrompt AABusinessClosed.wav parameter holidayPrompt AAHolidayPrompt.wav parameter welcomePrompt AAWelcome.wav parameter disconnectAfterMenu false parameter dialByFirstName false parameter allowExternalTransfers false parameter MaxRetry 3 parameter dialByExtnAnytime false parameter busOpenPrompt AABusinessOpen.wav parameter businessSchedule systemschedule parameter dialByExtnAnytimeInputLength 4 parameter operExtn 0 end application ccn application ciscomwiapplication aa description ciscomwiapplication enabled maxsessions 6 script setmwi.aef parameter
Re: [OSL | CCIE_Voice] Frame-relay fragment question
I think not, the fragment size is related to the amount of data that can be placed on the wire in 10 ms which relates to line speed not CIR Graham Hopkins On 29 Jun 2010, at 15:17, Bo Gao bga...@gmail.com wrote: HQ-BR1 bandwidth is 384K, I have the following config: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust If I were to change the cir to 95% based on the QoS SNRD Then I would have: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 34800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust Question: Should I also change the frame-realy fragment from 480 to 456? Why? Thank you! Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Voice Hunt Group to Voicemail Vol2 Lab 9
Thanks Roger, that's what I thought, ext 1000 is an alternative number for 1003 on UC so it get to the right mailbox. I always like to get a second opinion when the PG doesn't match my solution, although as with all things Cisco there is often more than one way to solve a problem Graham On 28 Jun 2010, at 11:55, Roger Källberg roger.kallb...@cygate.se wrote: Hi Graham, The big differnce that I can see is that the first config example will actually never even hit voice register dn 3, it will go to VM directly from the hunt group. But the second config will first use the hunt group, then the voice register dn and from there it will go to VM. At some stage in the second call flow the original DNIS=1000, is lost, thats why the VM see the call as a direct call from ANI=5002. If DNIS=1000 would have been kept it should have been in the form of RDNIS, as per what the first debug shows as diversion. Regards Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: Graham Hopkins [ghopk...@wolf-rock.co.uk] Skickat: den 25 juni 2010 18:39 Till: CCIE Voice Maillist Ämne: [OSL | CCIE_Voice] Voice Hunt Group to Voicemail Vol2 Lab 9 Some odd behaviour here that I can't quite get my head around. Config 1 - Working voice register dn 3 number 1003 call-forward b2bua all 1600 mwi voice hunt-group 1 parallel final 1600 list 1001,1002 timeout 12 pilot 1000 Config 2 - As per PG voice register dn 3 number 1003 call-forward b2bua all 1600 mwi voice hunt-group 1 parallel final 1003 list 1001,1002 timeout 12 pilot 1000 Call from 5002 to 1000 and let go to voicemail with config 1 UC sees the call as redirected and plays the welcome 1000 not available with config 2 UC see the call as coming directly from 5002 to 1600 and prompts for the pin for mailbox 5002 difference seems to be in the SIP messaging for config 1 I see Jun 25 13:14:44.778: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:1...@10.10.210.13:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.201.1:5060;branch=z9hG4bK10AD7 Remote-Party-ID: HQ Ph2 sip:5...@10.10.201.1;party=calling;screen=no;privacy=off From: HQ Ph2 sip:5...@10.10.201.1;tag=82568C-1CF To: sip:1...@10.10.210.13 Date: Fri, 25 Jun 2010 13:14:44 GMT Call-ID: 74e861cb-7f9211df-836ffb0f-d478...@10.10.201.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 1921735851-2140279263-2154205609-2551626315 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1277471684 Contact: sip:5...@10.10.201.1:5060 Diversion: sip:1...@10.10.201.1;privacy=off;reason=no-answer;counter=1;screen=no Expires: 180 Allow-Events: telephone-event Max-Forwards: 69 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 188 but no such diversion for config 2 just Jun 25 13:15:34.588: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:1...@10.10.210.13:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.201.1:5060;branch=z9hG4bK110200E Remote-Party-ID: HQ Ph2 sip:5...@10.10.201.1;party=calling;screen=no;privacy=off From: HQ Ph2 sip:5...@10.10.201.1;tag=831920-E7 To: sip:1...@10.10.210.13 Date: Fri, 25 Jun 2010 13:15:34 GMT Call-ID: 929827af-7f9211df-8380fb0f-d478...@10.10.201.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2459285263-2140279263-2206071567-222790417 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1277471734 Contact: sip:5...@10.10.201.1:5060 Expires: 180 Allow-Events: telephone-event Max-Forwards: 69 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 187 Any thoughts? Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME CUCM via CUBE
Mouhammad, Setup Lab 8 again and tested with the inter-cluster trunk - all worked without issue. Had faststart inbound/outbound at the trunk end on CUCM and a software MTP configured on the HQ router and registered to CUCM. I know that doesn't help your specific setup, but it may give you an idea where to look Regards Graham Hopkins On 27 Jun 2010, at 17:32, Graham Hopkins wrote: Interesting - when I did this lab I used a H.323 Gateway as well. This Cisco configuration note here http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml suggests There are two methods of defining an H.323 trunk to the Cisco Unified Border Element on the Cisco Unified Communications Manager: With a gatekeeper—Configure an H.225 trunk (GK controlled) toward Cisco Unified Border Element Without a gatekeeper—Configure the Cisco Unified Border Element as an H.323 gateway However I note that the Solution Guide has a non-GK controlled inter-cluster trunk which I guess you were testing. DIdn't do that myself and only looked at the solution guide today. Something to test further, however I can find no Cisco documents describing this as a supported solution. Anyone know where to find this? Regards Graham Hopkins On 27 Jun 2010, at 15:54, Mouhammad Nasser wrote: Hi there, Well, I wasn't able to receive the call on CUCM through the created trunk, so I thought of creating an H.323 gateway to receive calls through: it is working now, but with some configuration that I am not able to verify :( , as well as what is stated earlier, I added - Configure interface on HQ router as an h323-gateway voip interface - Add an H.323 gateway to CUCM with the following: - MRGL: HQ-MRGL, which contains hardware transcoder on HQ VG - No MTP Required - Wait for far end h.245 TCS - Inbound fast start enabled I think I am loosing the sense of the question here, while my only logic I followed was testing all possibilities, finally the above workd, but I hope if I can have an explaination Regards, From: ghopk...@wolf-rock.co.uk To: engnasse...@hotmail.com Subject: Re: [OSL | CCIE_Voice] CME CUCM via CUBE Date: Sun, 27 Jun 2010 12:50:07 +0100 CC: ccie_voice@onlinestudylist.com First need to decide if it's a signalling or a capabilities negotiation issue. Does the CUCM phone ring I presume not as you get unknown number ? Is CSS for the trunk from CUBE set correctly, look at what happens to the call setup after it leaves CUBE Graham On 27 Jun 2010, at 09:54, Mouhammad Nasser engnasse...@hotmail.com wrote: Hello everyone, I am working on Lab 8 to make calls between CUCM and CME through CUBE; anyway: I am using SCCP phones in CME instead of SIP. Now, I can call from HQ to CME, but calls from CME to HQ fails with error code 38, and error message on phone is Unknown number, although calls hit inbound and outgound dial-peers approperiately,can anyone gives any suggestion? P.S. MTP and fast start on CUCM trunk make no difference, I think they are required only if SIP endpoints are used, right? My configuration = CME == voice service voip allow-connections h323 to sip allow-connections sip to h323 sip bind control source-interface Loopback0 bind media source-interface Loopback0 ! ! ! dial-peer voice 100 voip description *** calls to CUCM through CUBE *** destination-pattern [15]... session protocol sipv2 session target ipv4:142.1.64.254 dtmf-relay rtp-nte no vad ! HQ === voice service voip allow-connections h323 to sip allow-connections sip to h323 sip bind control source-interface Loopback0 bind media source-interface Loopback0 ! ! ! dial-peer voice 100 voip description *** calls to CME *** destination-pattern 3... session protocol sipv2 session target ipv4:142.1.66.254 dtmf-relay rtp-nte no vad ! dial-peer voice 101 voip description *** inbound dial-peer for incoming CME calls *** session protocol sipv2 incoming called-number [15]... dtmf-relay rtp-nte ! dial-peer voice 102 voip description *** outbound dial-peer for incoming CME calls, points to CUCM *** destination-pattern [15]... session target ipv4:10.1.200.21 dtmf-relay h245-alphanumeric no vad ! ! sip-ua retry invite 2 timers trying 300 Thanks in advance Mouhammad Hotmail: Free, trusted and rich email service. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ For more information
Re: [OSL | CCIE_Voice] CME CUCM via CUBE
First need to decide if it's a signalling or a capabilities negotiation issue. Does the CUCM phone ring I presume not as you get unknown number ? Is CSS for the trunk from CUBE set correctly, look at what happens to the call setup after it leaves CUBE Graham On 27 Jun 2010, at 09:54, Mouhammad Nasser engnasse...@hotmail.com wrote: Hello everyone, I am working on Lab 8 to make calls between CUCM and CME through CUBE; anyway: I am using SCCP phones in CME instead of SIP. Now, I can call from HQ to CME, but calls from CME to HQ fails with error code 38, and error message on phone is Unknown number, although calls hit inbound and outgound dial-peers approperiately,can anyone gives any suggestion? P.S. MTP and fast start on CUCM trunk make no difference, I think they are required only if SIP endpoints are used, right? My configuration = CME == voice service voip allow-connections h323 to sip allow-connections sip to h323 sip bind control source-interface Loopback0 bind media source-interface Loopback0 ! ! ! dial-peer voice 100 voip description *** calls to CUCM through CUBE *** destination-pattern [15]... session protocol sipv2 session target ipv4:142.1.64.254 dtmf-relay rtp-nte no vad ! HQ === voice service voip allow-connections h323 to sip allow-connections sip to h323 sip bind control source-interface Loopback0 bind media source-interface Loopback0 ! ! ! dial-peer voice 100 voip description *** calls to CME *** destination-pattern 3... session protocol sipv2 session target ipv4:142.1.66.254 dtmf-relay rtp-nte no vad ! dial-peer voice 101 voip description *** inbound dial-peer for incoming CME calls *** session protocol sipv2 incoming called-number [15]... dtmf-relay rtp-nte ! dial-peer voice 102 voip description *** outbound dial-peer for incoming CME calls, points to CUCM *** destination-pattern [15]... session target ipv4:10.1.200.21 dtmf-relay h245-alphanumeric no vad ! ! sip-ua retry invite 2 timers trying 300 Thanks in advance Mouhammad Hotmail: Free, trusted and rich email service. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME CUCM via CUBE
Interesting - when I did this lab I used a H.323 Gateway as well. This Cisco configuration note here http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml suggests There are two methods of defining an H.323 trunk to the Cisco Unified Border Element on the Cisco Unified Communications Manager: With a gatekeeper—Configure an H.225 trunk (GK controlled) toward Cisco Unified Border Element Without a gatekeeper—Configure the Cisco Unified Border Element as an H.323 gateway However I note that the Solution Guide has a non-GK controlled inter-cluster trunk which I guess you were testing. DIdn't do that myself and only looked at the solution guide today. Something to test further, however I can find no Cisco documents describing this as a supported solution. Anyone know where to find this? Regards Graham Hopkins On 27 Jun 2010, at 15:54, Mouhammad Nasser wrote: Hi there, Well, I wasn't able to receive the call on CUCM through the created trunk, so I thought of creating an H.323 gateway to receive calls through: it is working now, but with some configuration that I am not able to verify :( , as well as what is stated earlier, I added - Configure interface on HQ router as an h323-gateway voip interface - Add an H.323 gateway to CUCM with the following: - MRGL: HQ-MRGL, which contains hardware transcoder on HQ VG - No MTP Required - Wait for far end h.245 TCS - Inbound fast start enabled I think I am loosing the sense of the question here, while my only logic I followed was testing all possibilities, finally the above workd, but I hope if I can have an explaination Regards, From: ghopk...@wolf-rock.co.uk To: engnasse...@hotmail.com Subject: Re: [OSL | CCIE_Voice] CME CUCM via CUBE Date: Sun, 27 Jun 2010 12:50:07 +0100 CC: ccie_voice@onlinestudylist.com First need to decide if it's a signalling or a capabilities negotiation issue. Does the CUCM phone ring I presume not as you get unknown number ? Is CSS for the trunk from CUBE set correctly, look at what happens to the call setup after it leaves CUBE Graham On 27 Jun 2010, at 09:54, Mouhammad Nasser engnasse...@hotmail.com wrote: Hello everyone, I am working on Lab 8 to make calls between CUCM and CME through CUBE; anyway: I am using SCCP phones in CME instead of SIP. Now, I can call from HQ to CME, but calls from CME to HQ fails with error code 38, and error message on phone is Unknown number, although calls hit inbound and outgound dial-peers approperiately,can anyone gives any suggestion? P.S. MTP and fast start on CUCM trunk make no difference, I think they are required only if SIP endpoints are used, right? My configuration = CME == voice service voip allow-connections h323 to sip allow-connections sip to h323 sip bind control source-interface Loopback0 bind media source-interface Loopback0 ! ! ! dial-peer voice 100 voip description *** calls to CUCM through CUBE *** destination-pattern [15]... session protocol sipv2 session target ipv4:142.1.64.254 dtmf-relay rtp-nte no vad ! HQ === voice service voip allow-connections h323 to sip allow-connections sip to h323 sip bind control source-interface Loopback0 bind media source-interface Loopback0 ! ! ! dial-peer voice 100 voip description *** calls to CME *** destination-pattern 3... session protocol sipv2 session target ipv4:142.1.66.254 dtmf-relay rtp-nte no vad ! dial-peer voice 101 voip description *** inbound dial-peer for incoming CME calls *** session protocol sipv2 incoming called-number [15]... dtmf-relay rtp-nte ! dial-peer voice 102 voip description *** outbound dial-peer for incoming CME calls, points to CUCM *** destination-pattern [15]... session target ipv4:10.1.200.21 dtmf-relay h245-alphanumeric no vad ! ! sip-ua retry invite 2 timers trying 300 Thanks in advance Mouhammad Hotmail: Free, trusted and rich email service. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Voice Hunt Group to Voicemail Vol2 Lab 9
Some odd behaviour here that I can't quite get my head around. Config 1 - Working voice register dn 3 number 1003 call-forward b2bua all 1600 mwi voice hunt-group 1 parallel final 1600 list 1001,1002 timeout 12 pilot 1000 Config 2 - As per PG voice register dn 3 number 1003 call-forward b2bua all 1600 mwi voice hunt-group 1 parallel final 1003 list 1001,1002 timeout 12 pilot 1000 Call from 5002 to 1000 and let go to voicemail with config 1 UC sees the call as redirected and plays the welcome 1000 not available with config 2 UC see the call as coming directly from 5002 to 1600 and prompts for the pin for mailbox 5002 difference seems to be in the SIP messaging for config 1 I see Jun 25 13:14:44.778: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:1...@10.10.210.13:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.201.1:5060;branch=z9hG4bK10AD7 Remote-Party-ID: HQ Ph2 sip:5...@10.10.201.1;party=calling;screen=no;privacy=off From: HQ Ph2 sip:5...@10.10.201.1;tag=82568C-1CF To: sip:1...@10.10.210.13 Date: Fri, 25 Jun 2010 13:14:44 GMT Call-ID: 74e861cb-7f9211df-836ffb0f-d478...@10.10.201.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 1921735851-2140279263-2154205609-2551626315 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1277471684 Contact: sip:5...@10.10.201.1:5060 Diversion: sip:1...@10.10.201.1;privacy=off;reason=no-answer;counter=1;screen=no Expires: 180 Allow-Events: telephone-event Max-Forwards: 69 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 188 but no such diversion for config 2 just Jun 25 13:15:34.588: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:1...@10.10.210.13:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.201.1:5060;branch=z9hG4bK110200E Remote-Party-ID: HQ Ph2 sip:5...@10.10.201.1;party=calling;screen=no;privacy=off From: HQ Ph2 sip:5...@10.10.201.1;tag=831920-E7 To: sip:1...@10.10.210.13 Date: Fri, 25 Jun 2010 13:15:34 GMT Call-ID: 929827af-7f9211df-8380fb0f-d478...@10.10.201.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2459285263-2140279263-2206071567-222790417 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1277471734 Contact: sip:5...@10.10.201.1:5060 Expires: 180 Allow-Events: telephone-event Max-Forwards: 69 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 187 Any thoughts? Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HWIC-4ESW POE issue
BTW adding the PoE daughter card will do no good unless you have the PoE version of the power supply in the router as well Graham On 24 Jun 2010, at 20:54, Adam Thompson phoe...@fatturtle.com wrote: If you just have a HWIC-4ESW, you won't have PoE. In order to have PoE you needs to have a HWIC-4ESW-POE module. You can try all you want, but you will never be able to enable PoE with a HWIC-4ESW module. Ref: http://www.cisco.com/en/US/prod/collateral/routers/ps5853/product_data_sheet0900aecd8016bf0b_ps5855_Products_Data_Sheet.html On Thu, Jun 24, 2010 at 3:45 PM, Deepak sidana sidana_dee...@yahoo.com wrote: Hi All, I am in process of setting home lab, i am facing a problem with HWIC-4ESW. I have crated L2 L3 vlans(vlan 200,400) on BR-II Router. configure the ports with following commands. switchport trunk native vlan 200 switchport mode trunk switchport voice vlan 400 i have also tried pppoe enable command under interface, but its also not helpul. When i plug the IP Phone, it does't recive the power,so vlan does't come up.If i plug my laptop driectly to port it work fine. Please suggest how to make the HWIC-4ESW Power enabled. BR Deepak ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP phones for CME
What does debug show Debug voice register error debug voice register events debug tftp events Regards Graham Hopkins On 21 Jun 2010, at 13:19, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: Hi guys, I'm working on lab9 Vol2, and I have 7961 phones registred to SIP CME, but every time the phones unregistred and registred again. Any Ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUCME Unicast MoH
Section in Vol 2 Lab 9 MoH from the CUCME routers (in my own lab) BR2 - multicast - fine to phones and PSTN BR1 - unicast - fine to phones and PSTN, phones are SIP and prefer G729, so transcoder in use. HQ - unicast - fine to PSTN but not to phones, default ephone seems to have multicast-moh, so have turned that off any ideas before I resort to Wireshark ? ephone 1 no multicast-moh device-security-mode none description HQ Phone1 mac-address 0024.14B3.662C type 7965 button 1:1 HQ-RTR#sh telephony-service ephone Number of Configured ephones 2 (Registered 2) ephone 1 Device Security Mode: Non-Secure mac-address 0024.14B3.662C type 7965 button 1:1 keepalive 30 auxiliary 30 max-calls-per-button 8 busy-trigger-per-button 0 Always send media packets to this router: No Preferred codec: g711ulaw conference drop-mode never conference add-mode all conference admin: No privacy: Yes privacy button: No user-locale US network-locale US Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCME Unicast MoH
Thanks Angel, nice question in Lab 9 - asking for something that doesn't work :-) Graham On 21 Jun 2010, at 18:18, Angel Perez wrote: Hi: Unicast is not permited beetween sccp phones for CME (thanks Amy), so no need for Whireshark :) you can only test uni from pstn thx From: ghopk...@wolf-rock.co.uk Date: Mon, 21 Jun 2010 18:11:54 +0100 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUCME Unicast MoH Section in Vol 2 Lab 9 MoH from the CUCME routers (in my own lab) BR2 - multicast - fine to phones and PSTN BR1 - unicast - fine to phones and PSTN, phones are SIP and prefer G729, so transcoder in use. HQ - unicast - fine to PSTN but not to phones, default ephone seems to have multicast-moh, so have turned that off any ideas before I resort to Wireshark ? ephone 1 no multicast-moh device-security-mode none description HQ Phone1 mac-address 0024.14B3.662C type 7965 button 1:1 HQ-RTR#sh telephony-service ephone Number of Configured ephones 2 (Registered 2) ephone 1 Device Security Mode: Non-Secure mac-address 0024.14B3.662C type 7965 button 1:1 keepalive 30 auxiliary 30 max-calls-per-button 8 busy-trigger-per-button 0 Always send media packets to this router: No Preferred codec: g711ulaw conference drop-mode never conference add-mode all conference admin: No privacy: Yes privacy button: No user-locale US network-locale US Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Problem Connection between HQ and BR1 (vol2 Lab8)
Is this on your on kit or one of the PL racks ? What is the status of the frame relay PVCs ? sh frame-relay pvc sh frame-relay pvc 101 etc and the lmi to the frame switch sh frame-relay lmi check that the status messages are being sent and received thus LMI Statistics for interface Serial1/0 (Frame Relay DTE) LMI TYPE = ANSI Invalid Unnumbered info 0 Invalid Prot Disc 0 Invalid dummy Call Ref 0 Invalid Msg Type 0 Invalid Status Message 0 Invalid Lock Shift 0 Invalid Information ID 0 Invalid Report IE Len 0 Invalid Report Request 0 Invalid Keep IE Len 0 Num Status Enq. Sent 18 Num Status msgs Rcvd 18 Num Update Status Rcvd 0 Num Status Timeouts 0 Last Full Status Req 00:00:06 Last Full Status Rcvd 00:00:06 Regards Graham On 20 Jun 2010, at 14:23, naoufal.kerboute wrote: Hi, I've a connection issue between HQ and BR1, I can't bring the interface dlci 201 up. below my configuration: BR1: interface Serial0/0/1:0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ! router ospf 1 router-id 10.10.101.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! HQ: interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! Also I tried to revert configuration on routers and do everything from start, but still have problem between HQ and BR1. After reconfigure everything connection BR2 and HQ works great. Any idea Thank you Naoufal ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge Not Working (Lab7 Vol2)
Do all devices have MRGLs that can see the bridge ? Also check privacy settings but looks like they are OK if remote in use shows uo Graham On 19 Jun 2010, at 14:51, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: Hi, I'm working on lab7 Vol2 section DISA dialing, And I can't get the cbarge to work. I've configured the single button Cbarge under the BR2Phone2, also the HW conf bridge on the BR2 GW registred to the CUCM, but when I call the HQ or BR1 phones from the BR2 Mobile Phones and answer the call, I can see on BR2Phone2 that is in remote in use but when I press the line button the phone display to conference but I here a busy tone. Any Idea? Thank you guys ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RE : RE : CBarge Not Working (Lab7 Vol2)
My understanding is that all legs of the call use the conference bridge, so when br2phone2 is talking to the conference bridge then the other two phones involved in the barge need to be talking to it as well. So that includes say the HQ Phone and the gateway which terminates the PSTN call. Barge would use the built in bridge on the phone but cbarge requires all call legs to suspend and reconnect to the bridge. Graham On 19 Jun 2010, at 17:54, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: Any Idea guys Message d'origine De: ccie_voice-boun...@onlinestudylist.com de la part de naoufal.kerboute Date: sam. 6/19/2010 2:03 À: Graham Hopkins Cc: ccie_voice@onlinestudylist.com Objet : [OSL | CCIE_Voice] RE : CBarge Not Working (Lab7 Vol2) I'v assigned only the BR2 phones to the mrgl, because I want to use the cbarge function only on bR2phon2 Message d'origine De: Graham Hopkins [mailto:ghopk...@wolf-rock.co.uk] Date: sam. 6/19/2010 1:58 À: naoufal.kerboute Cc: ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] CBarge Not Working (Lab7 Vol2) Do all devices have MRGLs that can see the bridge ? Also check privacy settings but looks like they are OK if remote in use shows uo Graham On 19 Jun 2010, at 14:51, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: Hi, I'm working on lab7 Vol2 section DISA dialing, And I can't get the cbarge to work. I've configured the single button Cbarge under the BR2Phone2, also the HW conf bridge on the BR2 GW registred to the CUCM, but when I call the HQ or BR1 phones from the BR2 Mobile Phones and answer the call, I can see on BR2Phone2 that is in remote in use but when I press the line button the phone display to conference but I here a busy tone. Any Idea? Thank you guys ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice #26244
Well done. It's always good to see someone's hard work rewarded Graham On 18 Jun 2010, at 19:46, Ashar Siddiqui siddas...@gmail.com wrote: Hello all, I went to Brussels yesterday and just an hour before learned that I am now officially CCIE Voice. It was my 2nd attempt but it was worth it. I learned a lot from my first attempt and it helped me build a better strategy for the 2nd. I am thankful to this wonderful list and IPExpert material which I used. Special thanks to Amy Ryan for her help whenever I needed. I am also grateful to my Study Partner Iwan Hoogendoorn, a triple CCIE and I was so lucky to have him as Study partner. I will never forget the way he use to make daily schedules and strictly made me follow those otherwise I am a lazy man..this number is for you Iwan! Few take home points for all those who will be making an attempt in coming days: 1 - Read the lab CAREFULLY (I made it Caps for a reason)..every word in a question is there for a reason! 2 - Do not rush! the mistakes you will make in first one hour will haunt you in the entire lab (unless you are lucky to figure out what went wrong) 3 - Do not spend too much time if something is not working - you can always come back to it. 4 - Note down sections and task which you are working and cross them as soon as you have completed it 5 - Call routing - This is how I did it, not necessarily helpful for you, I did call routing on a page first as what I am going to do at RL level, Pattern level etc..I configured everything first and then tested it one by one..took me 30 minutes to finish call routing 6 - Test everything you have done at least twice and as if it was configured by someone else and you are the proctor..I found one mistake while doing my 2nd check 7 - Save your config often, make sure before you leave that all gateways are up and registered to CUCM. I joined this list for my CCIE studies when I started my CCIE journey back in December 2009 but now I have decided to stick with it as I won't find such a nice bunch of people anywhere.. N.B: Above all, I loved my number..Digit '4' is my lucky number and Cisco made sure that I have enough of them.. :) Thank you all. It's party time now ;) Ashar Siddiqui CCIE#26244 (Voice) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 8 Question 4.3 - Live Record from Cell Phone
Correct, did that yesterday - must be working from an old lab will look for updates Regards Graham Hopkins On 18 Jun 2010, at 21:20, Matthew Berry ciscovoiceg...@gmail.com wrote: All - I have an older version of lab 8 that requests the ability to issue a live record session from a mobile phone call. This has since been removed from the Proctor Guide. Even so, I've been thinking about how such a request could be completed. In my mind, you'd just need to make the cell phone an alternate extension and conference in the live record DID from the cell phone. Does that sound right? Matthew ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] clock summer-time
But as Angel says these differ around the world and in some cases including the US have changed recently. For a lab I don' t hink its reasonable to expect people to know what these dates are I doubt that they can be looked up in the lab. Graham Hopkins On 9 Jun 2010, at 18:41, Ashar Siddiqui siddas...@gmail.com wrote: Oh Sorry I didn't understand your question initially..I thought you are asking about some command start/stop which I never did. If you are asking about recurring thing and then adding time as when to start and when to stop then YES I do it all the time. hth Angel Perez wrote: Hi: In real live thats depend on the timezone, for US time zones (PDT, EDT, ...) is not necessary becouse the default has the correct date, but for example at Europe summer time start at different week depending on the zone so you should manually configure. In the lab I suppose that you should ask proctor hth From: siddas...@gmail.com To: ciscovoiceg...@gmail.com; ccie_voice@onlinestudylist.com Date: Wed, 9 Jun 2010 16:36:00 +0100 Subject: Re: [OSL | CCIE_Voice] clock summer-time I have never done start/stop and it use to work fine. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Matthew Berry Sent: 09 June 2010 16:04 To: OSL Group Subject: [OSL | CCIE_Voice] clock summer-time Is it necessary to define a start/stop for the clock summer-time recurring command? I have been entering this as a general practice for all my exercises. However, I'm not sure if it's required to enter a start/stop time. Comments? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 Hotmail: Free, trusted and rich email service. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity Connection integration
Used to be the case that you had to manually add the servers but not now. I'm not sure in which version that changed. Graham Hopkins On 7 Jun 2010, at 03:59, Matthew Berry ciscovoiceg...@gmail.com wrote: I am comparing the IPexpert material to other documentation. In other documentation, there are references to adding Unity Connection as an application server under SYSTEM APPLICATION SERVER. The reason stated is so that CUC can obtain AXL access from CUCM. I cannot verify right now, but I don't recall ever setting this up on Proctor Lab's vracks. Can anyone advise? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol 2 Lab 7 3.2 MVA
Odd behaviour here - could well be my config of course but just wanted to see if anyone has come across this issue. Idea is to show the mobile phone number as the from number when calling in rather than the internal extension remote destination profile mobile +447976852817 linked to ext 3002 rdp css is set to css-rdp and MVA is se to use device css rather than gateway css. pt-internal contains extension 5002 pt-snr-3002 contains a translation pattern for 5002 with use external number mask When css-rdp contains only pt-snr-3002 For inward calls translation pattern is matched and and 5002 displays from +447976852817 but with MVR calls fails your call cannot be completed as dialled When css-rdp contains only pt-internal dn is matched and 5002 displays from 3002 MVR works fine. However if css-rdp contains pt-snr-3002 pt-internal then both direct calls and MVR work and and 5002 displays from +447976852817 So appears the translation pattern is only matched on MVR when the css contains another partition ! Any ideas Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP TRUNK
Yes you should pick it up in the invite and OK messages thus m=audio 47100 RTP/AVP 8 0 18 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:98 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes Regards Graham Hopkins On 29 May 2010, at 19:08, Brian Valentine wrote: You should try debug ccsip messages on the PSTN or CUBE router. It will show you the codec negotiation. On May 29, 2010 1:55 PM, Angel Perez gorr...@hotmail.com wrote: Hi: I have a sip trunk to my pstn router I'm trying to check the codec that the call is using but I can't this info at ucm traces or pstn gw debugs. I have try sip stack traces at ucm and also deb ccsip all at pstn, but I can't this info Any suggestion? Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 Lab_1
Check the option 150 on the DHCP scope the the HQ LAN as well. The phone should also show you where it thinks it's tftp server is. Graham On 28 May 2010, at 19:46, Gregory Bonton g.bon...@comcast.net wrote: I ran in to something straight as I was working thru Vol2 Lab 1. I was attempting to auto register the phones at HQ and BR1. The phone on BR1 registered fine the one on HQ would not. When I went to the switch, I could see that it got an IP and I could log into that phone via the IP address (10.10.200.30) and it should be that it had a number 1004, but it would not show up in the CUCM GUI. Can anybody tell if I did something wrong? HQ-3750#show cdp n f1/0/2 d - Device ID: SEP00119378D84E Entry address(es): IP address: 10.10.200.30 Platform: Cisco IP Phone 7960, Capabilities: Host Phone Interface: FastEthernet1/0/2, Port ID (outgoing port): Port 1 Holdtime : 127 sec Version : P00308000900 advertisement version: 2 Duplex: full Power drawn: 6.300 Watts Management address(es): HQ-3750#ping 10.10.200.30 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.200.30, timeout is 2 seconds: ! Rtr config ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.11 ip helper-address 10.10.210.10 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 3750 interface FastEthernet1/0/2 switchport access vlan 10 switchport mode access switchport voice vlan 20 spanning-tree portfast nterface FastEthernet1/0/3 ! interface FastEthernet1/0/4 switchport access vlan 30 switchport mode access duplex half spanning-tree portfast ! nterface Vlan10 ip address 10.10.100.3 255.255.255.0 ! HQ-RTR#ping 10.10.200.30 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.200.30, timeout is 2 seconds: ! Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/4 ms HQ-RTR#ping 10.10.100.3 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.100.3, timeout is 2 seconds: ! Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/4 ms HQ-RTR#ping 10.10.200.10 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.200.10, timeout is 2 seconds: ind Phone where image001.gif image002.gif Device Name(Line)image003.gif Description Device Pool Extension Partition Device Protocol Status IP Address Copy Super Copy image004.gif SEP001794DFFBE0(1) Auto 1003 HQ 1003 SCCP Registered with 10.10.210.10 10.10.201.30 image005.gif image006.gif ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] WB 1 LAB 5C General question
Randall The key point is that the 7-digit ANI for call to 911 and the 10-digit ANI for other calls have to work together. The lab will be marked as a whole. The question is one that if you answer by taking the obvious solution then you will break something that you have done before. Regards Graham Hopkins On 24 May 2010, at 07:48, Randall Crumm wrote: OK, So how often does the proctor come by to verify? Would he come by and verify all of lab 5 steps and score each step? Thanks, Randall hieng [mailto:r.ochi...@mfient.com] Sent: Sunday, May 23, 2010 11:34 PM To: Randall Crumm Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] WB 1 LAB 5C General question Then break say question 5.3 requirement that when calling 911 it should be 7 digits ANI? You'll get zero point there. The target is to ensure that all the requirements spelt out in the question are met at the end of all those configurations you can do. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm Sent: Monday, May 24, 2010 8:57 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] WB 1 LAB 5C General question HI, In lab 5c we first start out br1 with 7 digit ani, then it moves to 10 and we have to add a lot of translation patterns to make the ani 10 digits. Why can we just adjust the calling transformation pattern to 10 digit? Can we just change the calling transformation pattern in the real lab? Thanks, Randall -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, May 23, 2010 9:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 51, Issue 129 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Not getting PLUS on my Phones (Ashar Siddiqui) 2. CME background image 7961 (Ashar Siddiqui) -- Message: 1 Date: Sun, 23 May 2010 15:29:55 +0100 From: Ashar Siddiqui siddas...@gmail.com Subject: Re: [OSL | CCIE_Voice] Not getting PLUS on my Phones To: Ehab Salem esa...@sigma-it.net Cc: ccie_voice@onlinestudylist.com Message-ID: 4bf93be3.8070...@gmail.com Content-Type: text/plain; charset=us-ascii An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100523/5e8d1bc 5/attachment.html -- Message: 2 Date: Sun, 23 May 2010 16:19:06 +0100 From: Ashar Siddiqui siddas...@gmail.com Subject: [OSL | CCIE_Voice] CME background image 7961 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 4bf9476a.9090...@gmail.com Content-Type: text/plain; charset=us-ascii An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100523/7e1cda6 6/attachment-0001.htm -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 51, Issue 129 *** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links
Would appear that the issue is with ip cef per destination load sharing. The two links are load shared but rsvp attempts to always use the same link between the two loopbacks of the gateways so traffic to 30.30.30.30 ( BR1 loopback) is load-shared HQ-GW#sh ip cef 30.30.30.30 internal 30.30.30.30/32, epoch 0, RIB[I], refcount 5, per-destination sharing sources: RIB feature space: IPRM: 0x00038000 ifnums: Serial1/0.1(16): 10.10.10.2 Serial1/0.3(18): 10.10.10.10 path 68452BE8, path list 6845191C, share 0/1, type attached nexthop, for IPv4 nexthop 10.10.10.2 Serial1/0.1, adjacency IP adj out of Serial1/0.1 66A19B40 path 68452CD0, path list 6845191C, share 1/1, type attached nexthop, for IPv4 nexthop 10.10.10.10 Serial1/0.3, adjacency IP adj out of Serial1/0.3 67CB9E80 output chain: loadinfo 669D93E4, per-session, 2 choices, flags 0003, 6 locks flags: Per-session, for-rx-IPv4 16 hash buckets 0 IP adj out of Serial1/0.1 66A19B40 1 IP adj out of Serial1/0.3 67CB9E80 2 IP adj out of Serial1/0.1 66A19B40 3 IP adj out of Serial1/0.3 67CB9E80 4 IP adj out of Serial1/0.1 66A19B40 5 IP adj out of Serial1/0.3 67CB9E80 6 IP adj out of Serial1/0.1 66A19B40 7 IP adj out of Serial1/0.3 67CB9E80 8 IP adj out of Serial1/0.1 66A19B40 but the specific route used by rsvp always takes one route May 23 09:19:11.847: RSVP-API: 11.11.11.11_29550-30.30.30.30_24948[0.0.0.0]: Processing PATH request [id=0x67AC3BD0]... HQ-GW# sh ip cef exact-route 11.11.11.11 30.30.30.30 11.11.11.11 - 30.30.30.30 = IP adj out of Serial1/0.1 Other traffic - for example the rdp streams between phones on 192.168.60.x (HQ) and 192.168.50.x (BR1) does load share but never gets the chance ! HQ-GW# sh ip cef exact-route 192.168.60.2 192.168.50.10 192.168.60.2 - 192.168.50.10 = IP adj out of Serial1/0.1 HQ-GW# sh ip cef exact-route 192.168.60.4 192.168.50.10 192.168.60.4 - 192.168.50.10 = IP adj out of Serial1/0.3 Regards Graham Hopkins On 22 May 2010, at 19:19, Graham Hopkins wrote: I think we mean the same thing although my use of the term call setup is probably not a good one - when the request for bandwidth for call setup is made with G729 then 40 kbps is requested - worse case bandwidth for a 10ms sample rate. After call established this drops to 24kbps leaving 40kbps available for the bandwidth request of the second call. I think this is more likely to be a routing issue as the router makes no attempt to request bandwidth on the second link Gateways and debug follow - btw the configs have some legacy stuff from other testing - this is the dynamips version rather than the physical one so are 7200s HQ-GW#sh run Building configuration... Current configuration : 4800 bytes ! ! Last configuration change at 15:55:33 BST Sat May 22 2010 ! NVRAM config last updated at 15:56:06 BST Sat May 22 2010 ! upgrade fpd auto version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ-GW ! boot-start-marker boot-end-marker ! logging message-counter syslog ! no aaa new-model clock timezone GMT 0 clock summer-time BST recurring last Sun Mar 1:00 last Sun Oct 1:00 clock calendar-valid ip source-route ip cef ! ! ip dhcp excluded-address 192.168.60.1 192.168.60.9 ip dhcp excluded-address 192.168.60.21 192.168.60.254 ! ip dhcp pool PHONES network 192.168.60.0 255.255.255.0 default-router 192.168.60.1 option 150 ip 192.168.60.2 ! ! no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! voice service voip fax protocol cisco h323 ras rrq dynamic prefixes ! ! ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! ! archive log config hidekeys ! ! class-map match-all VOIP match ip dscp ef class-map match-any CONTROL match ip dscp cs3 af31 class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority percent 70 class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default fair-queue policy-map WAN class VOIP priority percent 25 compress header ip rtp class CONTROL bandwidth percent 30 class class-default fair-queue ! ! ! ! ! interface Loopback0 ip address 11.11.11.11 255.255.255.255 h323-gateway voip interface h323-gateway voip id GK1 ipaddr 20.20.20.20 1719 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 11.11.11.11 ! interface FastEthernet0/0 ip address 192.168.60.1 255.255.255.0 duplex half speed 100 ! interface FastEthernet0/1 description to GK no ip address shutdown duplex auto speed auto ! interface Serial1/0 no ip address encapsulation frame-relay serial restart-delay 0
Re: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue
True, but then you have to remember to explicitly set the intra-region codec to that required by other parts of the lab - for example G.711 in HQ - otherwise you break that. Regards Graham Hopkins On 22 May 2010, at 07:33, Matthew Hall wrote: In my experience changing the call manager service parameters for default inter and intra region codecs to g729 causes this to work in both directions. As long as your dial-peer is g729 and the GK trunk is in a g729 region. Matt On May 21, 2010, at 12:06 PM, Graham Hopkins wrote: Matthew - I found that the call from HQ to BR is fine and shows 16kbps, its the call from BR2 to HQ that has the problem, so to complete the task just make the call in the right direction :-) Did you have the same issue ? Regards Graham Hopkins On 21 May 2010, at 16:50, Berry, Matthew J. wrote: That’s great to know. I burned a few hours last night on Proctor trying to get this to work. Hopefully we won’t be asked a question like that on the lab. According to my understanding, then, we cannot technically complete and get points for question 5.1 since it requires you to produce the “show gatekeeper calls” output listed in the question. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: Friday, May 21, 2010 10:44 AM To: Berry, Matthew J.; CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: RE: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue Also this, http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16188.html Roger Källberg Unified Communication Consultant Cygate AB From: Berry, Matthew J. [mailto:mjbe...@krollontrack.com] Sent: den 21 maj 2010 17:21 To: CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue All – I had an issue last night on Vol 2 Lab 2. I am sending calls from HQ (Region = HQ) to BR2 over my H.225 trunk (Region = GK). Region setting between HQ and GK specifies G.729. I have a transcoder registered on the BR2 router. When I call across the gatekeeper, my endpoints show G.729, but “show gatekeeper calls” shows 128kbps. Extremely odd. Does anyone have insight into this? Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.com www.krollontrack.com | www.kroll.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links
Has anyone got this working/had problems etc. I have two links with 96k allocated per link but the second call (both G711) gets Not Enough Bandwidth. routing is load-sharing O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 RSVP call agents are up and registered to CUCM. Any ideas ? interface Serial1/0.1 point-to-point bandwidth 384 ip address 10.10.10.1 255.255.255.252 frame-relay interface-dlci 101 ip rsvp bandwidth 96 end HQ-GW#sh run int s1/0.3 Building configuration... Current configuration : 155 bytes ! interface Serial1/0.3 point-to-point bandwidth 384 ip address 10.10.10.9 255.255.255.252 frame-relay interface-dlci 111 ip rsvp bandwidth 96 end HQ-GW# HQ-GW#sh ip rsvp interface interfacersvp allocated i/f max flow max sub max Se1/0ena 80K1158K1158K0 Se1/0.1 ena 80K96K 96K 0 Se1/0.3 ena 0 96K 96K 0 dspfarm profile 1 mtp codec pass-through codec g711ulaw rsvp maximum sessions software 8 associate application SCCP O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links
Matthew - the two interfaces S1/0.1 and S1/0.3 are parallel links to the same remote site 96 K is allocated on each of the two links, enough for one call per link. This is based on Vol 2 Lab 5 scenario, according to the proctor guide the first call should use S1/0.1 and the second S1/0.3 but I never get a call on the second link - even if the bandwidth is set to 500K ! The actual example in Vol2 Lab 5 was to allow 4 calls at G.729 and the solution allowed 64K per sub interface ( i.e. 24K plus 40K for call setup) however I could not get more than two calls between the sites in this instance Regards Graham Hopkins On 22 May 2010, at 15:45, Matthew Berry wrote: You are not allocating enough bandwidth for two G711 calls with RSVP. One at 96 (worst case) and one at 64. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 22, 2010, at 8:48 AM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: Has anyone got this working/had problems etc. I have two links with 96k allocated per link but the second call (both G711) gets Not Enough Bandwidth. routing is load-sharing O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 RSVP call agents are up and registered to CUCM. Any ideas ? interface Serial1/0.1 point-to-point bandwidth 384 ip address 10.10.10.1 255.255.255.252 frame-relay interface-dlci 101 ip rsvp bandwidth 96 end HQ-GW#sh run int s1/0.3 Building configuration... Current configuration : 155 bytes ! interface Serial1/0.3 point-to-point bandwidth 384 ip address 10.10.10.9 255.255.255.252 frame-relay interface-dlci 111 ip rsvp bandwidth 96 end HQ-GW# HQ-GW#sh ip rsvp interface interfacersvp allocated i/f max flow max sub max Se1/0ena 80K1158K1158K0 Se1/0.1 ena 80K96K 96K 0 Se1/0.3 ena 0 96K 96K 0 dspfarm profile 1 mtp codec pass-through codec g711ulaw rsvp maximum sessions software 8 associate application SCCP O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Location based RSVP over dual Frame Relay Links
, reason: Local application requested tear May 22 19:12:48.734: RSVP: 11.11.11.11_20064-30.30.30.30_19784[0.0.0.0]: Expiring Serial1/0.1 RESV state, reason: Local application requested tear (17:20064) May 22 19:12:48.734: RSVP: 30.30.30.30_19784-11.11.11.11_20064[0.0.0.0]: Expiring RESV state, reason: Local application requested tear May 22 19:12:48.734: RSVP: 30.30.30.30_19784-11.11.11.11_20064[0.0.0.0]: Expiring receiver host RESV state, reason: Local application requested tear (17:19784) May 22 19:12:48.734: RSVP: 30.30.30.30_19784-11.11.11.11_20064[0.0.0.0]: Expiring Serial1/0.1 RESV request state, reason: Local application requested tear May 22 19:12:48.734: RSVP: 30.30.30.30_19784-11.11.11.11_20064[0.0.0.0]: Sending ResvTear message to 10.10.10.2 May 22 19:12:48.734: RSVP: session 11.11.11.11_20064[0.0.0.0]: Outgoing ResvTear, I/F=Se1/0.1, Layer=IP, NHOP=10.10.10.2, Prerouted=N IP HDR: 10.10.10.1-10.10.10.2, TOS=0x00, Len=108, TTL=255, RA=N RSVP HDR: RRC=N, TTL=255, Len=88, Cksum=0x8074 May 22 19:12:48.738: RSVP: session [TBD] Incoming PathTear, I/F=Se1/0.1, Layer=IP On 22 May 2010, at 18:28, Matthew Berry wrote: Graham, According to my understanding, the 64 Kbps does not equal 24 Kbps for the call and 40 Kbps for setup. Instead, the RSVP reservation always calculates the incoming call at the worst-case scenario of 40 Kbps for a g.729 call. The remaining 24 Kbps is for call #2. I am not familiar with lab 5 so I can't speak to the load balanced links. Could you send your gateway configs and the debug ip RSVP messages? Happy labbing! Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 22, 2010, at 10:06 AM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: Matthew - the two interfaces S1/0.1 and S1/0.3 are parallel links to the same remote site 96 K is allocated on each of the two links, enough for one call per link. This is based on Vol 2 Lab 5 scenario, according to the proctor guide the first call should use S1/0.1 and the second S1/0.3 but I never get a call on the second link - even if the bandwidth is set to 500K ! The actual example in Vol2 Lab 5 was to allow 4 calls at G.729 and the solution allowed 64K per sub interface ( i.e. 24K plus 40K for call setup) however I could not get more than two calls between the sites in this instance Regards Graham Hopkins On 22 May 2010, at 15:45, Matthew Berry wrote: You are not allocating enough bandwidth for two G711 calls with RSVP. One at 96 (worst case) and one at 64. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 22, 2010, at 8:48 AM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: Has anyone got this working/had problems etc. I have two links with 96k allocated per link but the second call (both G711) gets Not Enough Bandwidth. routing is load-sharing O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 RSVP call agents are up and registered to CUCM. Any ideas ? interface Serial1/0.1 point-to-point bandwidth 384 ip address 10.10.10.1 255.255.255.252 frame-relay interface-dlci 101 ip rsvp bandwidth 96 end HQ-GW#sh run int s1/0.3 Building configuration... Current configuration : 155 bytes ! interface Serial1/0.3 point-to-point bandwidth 384 ip address 10.10.10.9 255.255.255.252 frame-relay interface-dlci 111 ip rsvp bandwidth 96 end HQ-GW# HQ-GW#sh ip rsvp interface interfacersvp allocated i/f max flow max sub max Se1/0ena 80K1158K1158K0 Se1/0.1 ena 80K96K 96K 0 Se1/0.3 ena 0 96K 96K 0 dspfarm profile 1 mtp codec pass-through codec g711ulaw rsvp maximum sessions software 8 associate application SCCP O E2 192.168.50.0/24 [110/20] via 10.10.10.10, 00:23:33, Serial1/0.3 [110/20] via 10.10.10.2, 00:23:33, Serial1/0.1 Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME: IP source Address [any-match] and [strict-match]
If you have auto registration enabled then phones can register from anywhere and you have no control. In order to prevent this you can : a) turn off auto registration b) limit the networks the phone must be on to auto-register Regards Graham Hopkins On 21 May 2010, at 17:50, Mahdi Mohood wrote: Then if I restrict the other phones from the registration what is the value of these Phones ? (phones not registered) and there is no other call processing agent to allow other phones to register with. --- On Fri, 5/21/10, Matthew Berry ciscovoiceg...@gmail.com wrote: From: Matthew Berry ciscovoiceg...@gmail.com Subject: Re: [OSL | CCIE_Voice] CME: IP source Address [any-match] and [strict-match] To: Mahdi Mohood forccievo...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Friday, May 21, 2010, 2:03 PM No. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 21, 2010, at 4:26 AM, Mahdi Mohood forccievo...@yahoo.com wrote: Thank you for your reply. Do you mean I have to use this if I have more than one CME and I need to restrict the registration of the phones ? --- On Fri, 5/21/10, Matthew Berry ciscovoiceg...@gmail.com wrote: From: Matthew Berry ciscovoiceg...@gmail.com Subject: Re: [OSL | CCIE_Voice] CME: IP source Address [any-match] and [strict-match] To: ccie_voice@onlinestudylist.com Date: Friday, May 21, 2010, 4:19 AM If you have a router with three different VLANS (i.e. different subnets), you could restrict phones on subnets 2 and 3 from registering with the CME sourced from an IP on subnet 1. This would rarely be used, but might be useful to restrict devices from registering. Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 5/20/2010 9:35 PM, Mahdi Mohood wrote: Hi all I tried to read about the difference between the two commands [any-match and strict-match] but I did not find the exact answer. I understood that we are using this command to allow or deny the registration of phones. I found this in the archive of on line study: Use the *any-match* keyword to instruct the router to permit Cisco IP phone registration even when the IP server address used by the phone does not match the IP source address. This option can be used to allow registration of Cisco IP phones on different subnets or those with different default DHCP routers or different TFTP server addresses. Use the* strict-match *keyword to instruct the router to reject Cisco IP phone registration attempts if the IP server address used by the phone does not exactly match the source address. By dividing the Cisco IP phones into groups on different subnets and giving each group different DHCP default-router or TFTP server addresses, this option can be used to restrict the number of Cisco IP phones allowed to register. I could not understand how the IP phone will register with CME regardless of the IP address? and what is the relation between this and subnets and DHCP servers. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue
Matthew - I found that the call from HQ to BR is fine and shows 16kbps, its the call from BR2 to HQ that has the problem, so to complete the task just make the call in the right direction :-) Did you have the same issue ? Regards Graham Hopkins On 21 May 2010, at 16:50, Berry, Matthew J. wrote: That’s great to know. I burned a few hours last night on Proctor trying to get this to work. Hopefully we won’t be asked a question like that on the lab. According to my understanding, then, we cannot technically complete and get points for question 5.1 since it requires you to produce the “show gatekeeper calls” output listed in the question. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: Friday, May 21, 2010 10:44 AM To: Berry, Matthew J.; CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: RE: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue Also this, http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16188.html Roger Källberg Unified Communication Consultant Cygate AB From: Berry, Matthew J. [mailto:mjbe...@krollontrack.com] Sent: den 21 maj 2010 17:21 To: CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue All – I had an issue last night on Vol 2 Lab 2. I am sending calls from HQ (Region = HQ) to BR2 over my H.225 trunk (Region = GK). Region setting between HQ and GK specifies G.729. I have a transcoder registered on the BR2 router. When I call across the gatekeeper, my endpoints show G.729, but “show gatekeeper calls” shows 128kbps. Extremely odd. Does anyone have insight into this? Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.com www.krollontrack.com | www.kroll.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Basic VMWare Server Lab Question
This has now moved to V8 https://cisco.mediuscorp.com/market/networkers/listSubCat.se.work?TRGT=10/nxt/rcrs/=1180 Even so if you are outside the US the postage is more than the kit - so as suggested contact your local reseller Regards Graham Hopkins On 20 May 2010, at 15:14, Pulos, Greg wrote: To get the software, contact your Cisco Rep and request. There used to be a site which is the Cisco Market Place to order your NFR (not for resale) Kit of the entire UC 7.01 platform; but this seems to show it is not available currently. (may require cco login) https://cisco.mediuscorp.com/market/networkers/listSubCat.se.work?TRGT=10/nxt/rcrs/=1180 Thank you. greg -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of amir.safa...@memorialhealthsystem.com Sent: Wednesday, May 19, 2010 4:12 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Basic VMWare Server Lab Question I am looking for the best approach to building a VMWare server for running CUCM, UCCX and UC. I have an IBM 3650 M2 with lots of CPU, Memory and drive space. \ What is the best OS and VMWare application to use as my foundation? Once that is built, what is the best way to get the installation media for 7.x versions of CUCM, UCCX and UC testing in our lab? We are a large enterprise customer and we have software subscriptions from Cisco. Often those subscriptions only provide an upgrade path and not the original installation media. We're running 4.x in production on all the above mentioned platforms and certainly don't want to perform original installations of 4.x and then use our software subscription to upgrade to 7.x Amir ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] cannot dial from MVA
Mobile Connect use the original ANI received from the H.323 GW without using Incoming Calling Party Settings at gw level to match RD? Yes, I can confirm that - just saw that on Vik's video yesterday and ran some tests myself - I was having problems with the ANI being different on different gateways, using the full number for the RD and partial match did the trick. Regards Graham Hopkins On 19 May 2010, at 13:42, Peter Farkas wrote: Thank you for the link however my case is different a bit. Finally I could step over by looking sdi traces in depth: H.323 gw put '+1' at the begining of the ANI to be in E.164 format but the RD was defined without that. The behaviour was strange to me since Mobile Connect works as expected so RD is reachable in this case. In the other hand a call from the remote destination can succesfully authenticate. I have a question: Mobile Connect use the original ANI received from the H.323 GW without using Incoming Calling Party Settings at gw level to match RD? - Original Message - From: Angel Perez To: wormh...@sch.hu ; osl osl Sent: Wednesday, May 19, 2010 9:10 AM Subject: RE: [OSL | CCIE_Voice] cannot dial from MVA Hi, check this topic: http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16572.html hth From: wormh...@sch.hu To: ccie_voice@onlinestudylist.com Date: Tue, 18 May 2010 20:24:30 +0200 Subject: [OSL | CCIE_Voice] cannot dial from MVA Gents, I have an issue with MVA. MVA collects PIN and I press 1 to dial but it does not proceed with any call instead the well known prompt sounds: The call cannot be completed... Even if the called number is local and placed in the None partition. This prompt suggests CSS issue however as Vik advised before I created a totally new CSS just for RDP but it does not solve the problem. Service Parameters: Complete Match and RDP+Line CSS. I have read near all the thread regarding MVA here, but the issue remains. I attached the vxml debug. Any suggestion? Hotmail: Powerful Free email with security by Microsoft. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Physical Components for CCIE Voice Lab
Well there are so many options, you can work with the online rack rentals from Proctorlabs which are designed to work with the IP Expert workbooks. I don't use them myself, but other here can tell you how they got on with them, or you can build your own lab as you suggest. If building your own lab then I'd take a look at the areas from the lab blueprint that you think you need to focus on and get some kit to help you do that. 2800 series routers are fine and less expensive than the 2900 series, You'll need more than one if you want to practice things such as WAN QoS and CAC mechanisms, simulating E1/T1 circuits with crossover cables, call routing with SRST and CUCME etc. You'll need some DSPs installed for transcoding and other hardware media resources. For some areas dynamips is a handy tool, hang some phones off USB-to-Ethernet Interfaces and you can easily build a multi-router network to test all the various gateways, gatekeepers, directory gatekeepers, CUBE, B-ACD, Frame Relay QoS etc. sure there are some limitations to consider - such as no DSP resources for transcoding, but you can do a lot and its easy to clone scenarios and wipe and reboot the routers (much faster than the real thing). 7942 phones are fine but the 7961/61/65 phones give you more buttons which allow you to do more on a single phone. Regards Graham Hopkins On 15 May 2010, at 20:28, amuno...@hotmail.com amuno...@hotmail.com wrote: Hello, I am recently passed the CCIE Voice Written, then I am so excited for going on with the CCIE Voice Lab. The question that I have for yours, what physical components such as router should I buy for preparing for the Lab??? I have thought in buying the following: · router C2901-CME-SRST/K9, included Unity Express base release 8.0 · Two ip phones 7942G · Server for virtualization of CUCM (Pub + Sub), UC, UCCX, UPS, WinXP for IP Communicator. What could you suggest me for preparing for the Lab??when I feel that I am ready, I will take a bootcamp with IPExpert. I would appreciate your help and experience in this case, I want to start well since the beginning. Best regards, Alexis Munoz CCNP, CCVP, PMP ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VOL2 LAB4 6.1 BACD PROBLEM
Try pointing the dial peer 3500 to the loopback 0 where h.323 is bound rather than the vlan interface. I think that is needed to work with the PSTN correctly. Regards Graham Hopkins On 8 May 2010, at 11:59, Tom wrote: hi, i am having issues with VOL2 LAB4 6.1 , when i call in from the pstn i hear the cue AA when i press 3500, it giving me long busy tones. But, I can call BACD directly from br2 phones and get through to the hunt group perfectly. Any Idea configuration on the r3 is given below. 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 sip ! ! ! ! ! ! ! ! ! ! ! ! ! voice hunt-group 1 parallel list 3001,3002 pilot 3210 ! ! ! voice translation-rule 1 rule 1 /.*\($\)/ /\1/ ! ! voice translation-profile 4digit translate called 1 ! voice translation-profile e164-ani translate calling 1 ! ! voice-card 0 dsp services dspfarm ! ! application service queue flash:app-b-acd-2.1.2.3.tcl param aa-hunt3 3210 param queue-len 10 param aa-hunt10 3002 param queue-manager-debugs 1 param number-of-hunt-grps 2 ! service Alternate Default ! service aa flash:app-b-acd-aa-2.1.2.3.tcl paramspace english index 1 param number-of-hunt-grps 2 param menu-timeout 6 param dial-by-extension-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3500 paramspace english location flash: param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param max-time-call-retry 90 param voice-mail 3600 param service-name queue ! ! ! ! ! ! archive log config hidekeys ! ! controller E1 0/1/0 channel-group 0 timeslots 1-24 ! controller E1 0/1/1 framing NO-CRC4 pri-group timeslots 1-3,16 service mgcp ! ! ! ! ! interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id ZONE_01 ipaddr 177.1.254.1 1719 h323-gateway voip h323-id BR2_GW h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 177.1.254.3 ! interface FastEthernet0/0 description == To SW2 no ip address duplex auto speed auto ! interface FastEthernet0/0.11 description == Voice VLAN encapsulation dot1Q 11 ip address 177.3.11.1 255.255.255.0 ! interface FastEthernet0/0.12 description == Data VLAN encapsulation dot1Q 12 ip address 177.3.12.1 255.255.255.0 ! interface FastEthernet0/1 ip address 192.168.1.5 255.255.255.0 ip nat outside ip virtual-reassembly shutdown duplex auto speed auto ! interface Serial0/1/0:0 description == Frame Relay no ip address encapsulation frame-relay ! interface Serial0/1/0:0.1 point-to-point description == To HQ ip address 177.0.201.2 255.255.255.0 ip access-group block-ccm in snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/1/1:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! interface Service-Engine1/0 ip unnumbered FastEthernet0/0.11 service-module ip address 177.3.11.254 255.255.255.0 service-module ip default-gateway 177.3.11.1 ! router ospf 1 log-adjacency-changes network 0.0.0.0 255.255.255.255 area 0 default-information originate ! ip forward-protocol nd ip route 177.3.11.254 255.255.255.255 Service-Engine1/0 ! ip http server no ip http secure-server ip http path flash: ! ! ip access-list extended block-ccm deny ip host 192.168.2.11 any deny ip host 192.168.2.12 any deny ip any host 192.168.2.11 deny ip any host 192.168.2.12 permit ip any any ! ! ! ! tftp-server flash:en_bacd_music_on_hold.au ! control-plane ! ! ! voice-port 0/0/0 ! voice-port 0/0/1 ! voice-port 0/0/2 ! voice-port 0/0/3 ! voice-port 0/1/1:15 ! ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host 192.168.2.11 ccm-manager mgcp ccm-manager fax protocol cisco ccm-manager music-on-hold ! mgcp mgcp call-agent 192.168.2.12 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface Loopback0 mgcp bind media source-interface Loopback0 ! mgcp profile default ! sccp local FastEthernet0/0.11 sccp ccm 192.168.2.12 identifier 1 version 7.0 sccp ccm 192.168.2.11 identifier 2 version 7.0 sccp ip precedence 3 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register br2-xcode ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 8 associate application SCCP ! ! dial-peer voice 1 pots
Re: [OSL | CCIE_Voice] IPMA and Intercom Line Labels
Just tested this and yes it does work either way. The only thing I found of interest here is that for the labels to change after altering the intercom settings on the manager/assistant screens a restart of the IPMA service is required. Graham Hopkins On 29 Apr 2010, at 20:27, vccie2010 wrote: thx Mathew, that helps... On Thu, Apr 29, 2010 at 11:46 AM, Matthew Berry ciscovoiceg...@gmail.com wrote: It does still work. I don't have time to validate right now. However, the PG shows how to set this up without associating the intercom line. Good luck! Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 4/29/2010 1:45 PM, vccie2010 wrote: Mathew, If you don't define Intercom under Manager configs , does your intercom still work in IPMA mode while both Mgr and Asst are logged in? Could you please validate for us. thx On Thu, Apr 29, 2010 at 11:38 AM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: Matthew, thanks for that. Graham On 29 Apr 2010, at 19:34, Matthew Berry ciscovoiceg...@gmail.com wrote: Graham - This is what I noticed during that lab. Under the Manager/Assistant configuration screen there is the option to define intercom numbers. If you leave that drop-down field as Unassigned, when you login to IPMA the name will not change. However, if you specify an intercom in that field, the label will change. Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 4/29/2010 12:54 PM, Graham Hopkins wrote: Some of the labs show an extension - say 1080 and an intercom say *1080 as labels on the phone. When I tested IPMA with 7960s this is what I got. However have since started using some 7961s and the line label changes. When the assistant is logged out of the desktop app label reads *1080 when they are logged in it changes to manager. Setting labels manually in the line config doesn't seem to make any difference. Really wanted to know of this relates to the phone model or if there is some other setting I have overlooked? Obviously the name label is better for users but I guess it doesn't fulfil the the workbook requirement. Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPMA and Intercom Line Labels
Matthew, thanks for that. Graham On 29 Apr 2010, at 19:34, Matthew Berry ciscovoiceg...@gmail.com wrote: Graham - This is what I noticed during that lab. Under the Manager/Assistant configuration screen there is the option to define intercom numbers. If you leave that drop-down field as Unassigned, when you login to IPMA the name will not change. However, if you specify an intercom in that field, the label will change. Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 4/29/2010 12:54 PM, Graham Hopkins wrote: Some of the labs show an extension - say 1080 and an intercom say *1080 as labels on the phone. When I tested IPMA with 7960s this is what I got. However have since started using some 7961s and the line label changes. When the assistant is logged out of the desktop app label reads *1080 when they are logged in it changes to manager. Setting labels manually in the line config doesn't seem to make any difference. Really wanted to know of this relates to the phone model or if there is some other setting I have overlooked? Obviously the name label is better for users but I guess it doesn't fulfil the the workbook requirement. Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2 5.1 Gatekeeper Bandwidth
Yes this bug displays the symptoms I am getting, bug toolkit claims it is fixed in 7.0(1.11000.2) - which is my version - still I'm sure the Cisco solution will to be upgrade :-) Regards Graham On 27 Apr 2010, at 11:09, Angel Perez wrote: Hi Graham: It is a bug: CSCsl74701 hth From: ghopk...@wolf-rock.co.uk Date: Tue, 27 Apr 2010 09:24:09 +0100 To: vccie2...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 2 5.1 Gatekeeper Bandwidth vccie thanks, Tthe BRQ Enabled makes no difference but the Intra-region Codec to G729 solved the issue. However I'm not 100% sure why but here's what I think - can you confirm? The gatekeeper trunk is in a region BR1 the HQ gateway is in a region HQ so calls from HQ to BR1 via the trunk are inter-region and request 16Kbps from the gatekeeper but calls from BR1 to HQ as seen as intra-region as they appear to come from the HQ gateway so request 128Kbps from the gatekeeper even though both endpoints are G.729 I had considered all these calls to be inter-region, but that's obviously not the case. Regards Graham Hopkins On 27 Apr 2010, at 01:16, vccie2010 wrote: Do you have CCM SP's BRQ Enabled set to True and Intra-region Codec to G729 On Mon, Apr 26, 2010 at 8:16 AM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: I have all calls working from HQ/BR1 to BR2 and from BR2 to HQ/BR1, the SCCP phones use G.729 and the SIP phones at BR2 G.711 transcoding is invoked both at BR2 when a SIP phone is involved. However calls to BR2 appear as 16 Kbps in the gatekeeper as per the question calls from BR2 appear as 128Kbps in the gatekeeper, even when the SSCP phones at each end show G.729 and no transcoding is in use. Any ideas ? How does the gatekeeper derive this bandwidth, is it taken from a setup request even if a lower speed codec is then negotiated ? Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2 5.1 Gatekeeper Bandwidth (vccie2010)
I tend to agree about not using global parameters to fix this. The trunk already is in such a region. The problem isn't that G.729 codecs aren't being used, but that calls from BR2 to HQ request the wrong amount of bandwidth from the gatekeeper. As Angel has pointed out this is a known bug. Regards Graham On 27 Apr 2010, at 15:17, sean hurricane wrote: I think a better solution would be to configure a G729 region that only speaks G729 to all other regions and device pool and put the gatekeeper trunk in the device pool that way all communications are sure to be 16kps. Changing service Parameter is global and it may cost you in other areas. if you change your intra region codec to G729, in essence you have turned your cluster to a G729 only cluster and it may affect things like MOH and others.. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol 2 Lab 2 5.1 Gatekeeper Bandwidth
I have all calls working from HQ/BR1 to BR2 and from BR2 to HQ/BR1, the SCCP phones use G.729 and the SIP phones at BR2 G.711 transcoding is invoked both at BR2 when a SIP phone is involved. However calls to BR2 appear as 16 Kbps in the gatekeeper as per the question calls from BR2 appear as 128Kbps in the gatekeeper, even when the SSCP phones at each end show G.729 and no transcoding is in use. Any ideas ? How does the gatekeeper derive this bandwidth, is it taken from a setup request even if a lower speed codec is then negotiated ? Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] (no subject)
Also worth taking a look at http://ciscounitytools.com/index.html the port status monitor there is pretty good - as are most of the other tools Ryan Schwab schwab...@shaw.ca Regards Graham Hopkins On 19 Mar 2010, at 04:49, Ryan Schwab wrote: Hi Jean, Yes, the RTMT is definitely what you need. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jean M. Thewissen Sent: March-18-10 10:07 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] (no subject) Anyone can point me to the best way to monitor unity connection ports? I need to see info like dnis, rdnis and the like when a call comes in. Is RTMT the right tool for this? Thx! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] (no subject)
Mike, are you sure ? See http://ciscounitytools.com/Applications/CxN/PortStatusMonitorCUC7x/PortStatusMonitorCUC7x.html I have this running with UC Regards Graham Hopkins On 19 Mar 2010, at 11:07, Mike Thompson wrote: He's talking about UC and not Unity. There's no port status monitor in UC which is REAL disappointing. Sent from my phone, apologies for any typos. On Mar 19, 2010, at 6:37 AM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: Also worth taking a look at http://ciscounitytools.com/index.html the port status monitor there is pretty good - as are most of the other tools Ryan Schwab schwab...@shaw.ca Regards Graham Hopkins On 19 Mar 2010, at 04:49, Ryan Schwab wrote: Hi Jean, Yes, the RTMT is definitely what you need. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jean M. Thewissen Sent: March-18-10 10:07 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] (no subject) Anyone can point me to the best way to monitor unity connection ports? I need to see info like dnis, rdnis and the like when a call comes in. Is RTMT the right tool for this? Thx! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX Issue
I encountered this problem and the debug showed I wasn't even hitting the script. However found a comment on the cisco learning network that in UCCX 7.0 scripts will not work if saved in the system default folder, even if you have edited one of the existing scripts - so I saved the script elsewhere and all was fine. Regards Graham Hopkins On 22 Jan 2010, at 06:44, Kevin Damisch wrote: Validating the script only does a syntax check, such as making sure you don't have any dangling goto steps for example. You can pass validation on a script, but have problems when it runs, such as corrupted wav files for example. Run a reactive debug, call the trigger, and see which step the script stops at when you get your error message. Thanks, Kevin From: vccie2010 [vccie2...@gmail.com] Sent: Thursday, January 21, 2010 10:21 PM To: Kevin Damisch Cc: Otto Sanchez; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX Issue Yes, I did validated the script after I opned the the default ICD script , saved as ICDTEST , validated it too but no luck. On Thu, Jan 21, 2010 at 7:29 PM, Kevin Damisch kevin.dami...@vitalsite.commailto:kevin.dami...@vitalsite.com wrote: If you do a reactive debug, which step is the script at when you hear the message? Thanks, Kevin From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 [vccie2...@gmail.commailto:vccie2...@gmail.com] Sent: Thursday, January 21, 2010 9:14 PM To: Otto Sanchez Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX Issue Yes Otto, I did that please but still having same issue. On Thu, Jan 21, 2010 at 1:13 PM, Otto Sanchez o...@ipexpert.commailto:o...@ipexpert.commailto:o...@ipexpert.commailto:o...@ipexpert.com wrote: Did you validated the script once it was saved with the new name?, a common cause for this error is that the queue name is invalid in the application section configuration, On Thu, Jan 21, 2010 at 1:37 AM, vccie2010 vccie2...@gmail.commailto:vccie2...@gmail.commailto:vccie2...@gmail.commailto:vccie2...@gmail.com wrote: Well I am just saving the defualt ICD script as a different name icdtest and the moment I call I get the error message I posted ealrier. I have all csq etc as taken care of as with default script it warks fine. On Wed, Jan 20, 2010 at 6:50 PM, kill mill jha...@gmail.commailto:jha...@gmail.commailto:jha...@gmail.commailto:jha...@gmail.com wrote: THis is a general issue you have to decode the script to see what the issue is. plus check which script you are referecing and all the parameters csq etc are in line On Wed, Jan 20, 2010 at 8:45 PM, vccie2010 vccie2...@gmail.commailto:vccie2...@gmail.commailto:vccie2...@gmail.commailto:vccie2...@gmail.com wrote: I have UCCX on vmware. I am able to make calls succesfully to UCCX when there is default ICD script selected, but once I open the default ICD script in CRS editor and rename that suppose as icdnew and upload and select i, now when I call the trigger it prompts Thank you for calling…I am sorry. We are currently experiencing system problem. Please try again later Does anyone had same issue or can guide me what could be the problem. The CRS editor was donwloaded from the UCCX server itself. Seems like somehow the CRS editor is not saving the .aef file properly or ??? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/http://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.comhttp://www.ipexpert.com/http://www.ipexpert.com/ This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may
Re: [OSL | CCIE_Voice] UCCX Issue
Just ran the same test: Opened icd.aef and saved it twice as icd-test.aef once back to C:\Program Files\wfavvid\Scripts\system\default and once to my documents. Then uploaded each to UCCX. The one from my documents works and the one from the system folder gives the i'm sorry we are experiencing a system problem.. Original thread here https://supportforums.cisco.com/message/2013159;jsessionid=65DB330840174ED2468DBAE21FD30962.node0 Regards Graham On 22 Jan 2010, at 17:38, vccie2010 wrote: And Thanks Grahm for your input, I will try to do this too and update. On Thu, Jan 21, 2010 at 9:23 PM, Tanner Ezell tanner.ez...@gmail.com wrote: You really need to reactively debug the script and find specifically where it is failing. That message is entirely generic. It could be because the name used for a prompt doesn't include .wav at the end, etc etc. Find the step, let us know. On Thu, Jan 21, 2010 at 11:21 PM, vccie2010 vccie2...@gmail.com wrote: Yes, I did validated the script after I opned the the default ICD script , saved as ICDTEST , validated it too but no luck. On Thu, Jan 21, 2010 at 7:29 PM, Kevin Damisch kevin.dami...@vitalsite.com wrote: If you do a reactive debug, which step is the script at when you hear the message? Thanks, Kevin From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 [vccie2...@gmail.com] Sent: Thursday, January 21, 2010 9:14 PM To: Otto Sanchez Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX Issue Yes Otto, I did that please but still having same issue. On Thu, Jan 21, 2010 at 1:13 PM, Otto Sanchez o...@ipexpert.commailto:o...@ipexpert.com wrote: Did you validated the script once it was saved with the new name?, a common cause for this error is that the queue name is invalid in the application section configuration, On Thu, Jan 21, 2010 at 1:37 AM, vccie2010 vccie2...@gmail.commailto:vccie2...@gmail.com wrote: Well I am just saving the defualt ICD script as a different name icdtest and the moment I call I get the error message I posted ealrier. I have all csq etc as taken care of as with default script it warks fine. On Wed, Jan 20, 2010 at 6:50 PM, kill mill jha...@gmail.commailto:jha...@gmail.com wrote: THis is a general issue you have to decode the script to see what the issue is. plus check which script you are referecing and all the parameters csq etc are in line On Wed, Jan 20, 2010 at 8:45 PM, vccie2010 vccie2...@gmail.commailto:vccie2...@gmail.com wrote: I have UCCX on vmware. I am able to make calls succesfully to UCCX when there is default ICD script selected, but once I open the default ICD script in CRS editor and rename that suppose as icdnew and upload and select i, now when I call the trigger it prompts Thank you for calling…I am sorry. We are currently experiencing system problem. Please try again later Does anyone had same issue or can guide me what could be the problem. The CRS editor was donwloaded from the UCCX server itself. Seems like somehow the CRS editor is not saving the .aef file properly or ??? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.comhttp://www.ipexpert.com/ This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Tanner Ezell ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call is not going via BR1 when HQ is down
Have you checked the state of the Stop Routing on Unallocated Number Flag ? Graham Hopkins On 21 Jan 2010, at 18:01, Arun Kumar wrote: Hi All, I've configured my Route List with HQ and BR1 as backup and as per lab to test this I've shutdown the HQ voice port but when I'm calling from HQ phone I don't see any call coming and I'm keep on getting reorder tone on the phone. Thanks Arun ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol 1 Lab 12 CTI Route Point and Ports Not Registering
UCCX and CUCM seem to be set up as per the PG. However although the route point and ports are created they do not register. Restarted all servers and CTIManager but nothing changes Any tips on what to try/test please ? Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MLPP and traffic shaping
Can you post the interface configurations. I've had some issues in this area and you do need to ensure all the templates and policies line up. For example I'm still investigating a situation where I get two cloned virtual-acess interfaces and only one has the service policy applied ! interface Virtual-Access1 bandwidth 768 ip address 10.10.112.1 255.255.255.0 end HQ-RTR#sh run int virtual-access 3 Building configuration... Current configuration : 117 bytes ! interface Virtual-Access3 bandwidth 768 ip address 10.10.112.1 255.255.255.0 service-policy output 768kbps end Regards Graham Hopkins On 5 Jan 2010, at 09:23, Omotayo wrote: Hello, i have configured MLP LFI between hq and br1 router. i reloaded the routers but i can not ping across( this i can do prior to the configuration) with show policy-map i have the following output HQ-RTR#sh policy-map Interface Virtual-Access1 Virtual-Access1 Service-policy output: policy Service policy policy is in suspended mode HQ-RTR#ping 10.10.201.1 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.201.1, timeout is 2 seconds: . Success rate is 0 percent (0/5) HQ-RTR# On the br1 router, i have the following BR1-RTR#sh policy-map Interface Virtual-Access1 Virtual-Access1 Service-policy output: policy queue stats for all priority classes: queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 Class-map: media (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp ef (46) 0 packets, 0 bytes 5 minute rate 0 bps Priority: 33% (126 kbps), burst bytes 3150, b/w exceed drops: 0 Class-map: control (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp cs3 (24) 0 packets, 0 bytes 5 minute rate 0 bps Match: ip dscp af31 (26) 0 packets, 0 bytes 5 minute rate 0 bps Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 bandwidth 5% (19 kbps) Class-map: class-default (match-any) 254 packets, 6482 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops/flowdrops) 250/3/0/3 (pkts output/bytes output) 251/6407 Fair-queue: per-flow queue limit 16 BR1-RTR# ping 10.10.200.3 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.200.3, timeout is 2 seconds: . Success rate is 0 percent (0/5) BR1-RTR# When I set the service-policy output , I get an error message Class Based Weighted Fair Queueing will be applied only to the Virtual-Access interfaces associated with an MLP bundle. Any ideas on what is wrong thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MLPP and traffic shaping
I'd try removing the IP address from interface Serial0/0/1:0.1 point-to-point on the HQ Router. Then take a look at sh ppp multilink interface virtual-access 1 to see if the link is coming up as expected Regards Graham Hopkins On 5 Jan 2010, at 11:52, Omotayo wrote: hostname HQ-RTR ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs-mz.124-20.T1.bin warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! vtp mode transparent archive log config hidekeys ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! class-map match-any CONTROL match ip dscp cs3 match ip dscp af31 class-map match-any RTP match ip dscp ef ! ! policy-map POLICY-CHECK class RTP priority percent 33 compress header ip rtp class CONTROL bandwidth percent 5 class class-default fair-queue ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown 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 no ip address encapsulation frame-relay frame-relay traffic-shaping frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template200 class traffic-shape ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.1 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output POLICY-CHECK ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! ! no ip http server ip http authentication local no ip http secure-server ! ! map-class frame-relay traffic-shape frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 hostname BR1-RTR ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs-mz.124-20.T1.bin warm-reboot boot-end-marker ! logging message-counter syslog ! no aaa new-model memory-size iomem 20 clock timezone cst -6 clock summer-time cst recurring network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 ! dot11 syslog ip source-route ! ! ip cef ! ! ip domain name proctorlabs.com no ipv6 cef ! multilink bundle-name authenticated ! ! isdn switch-type primary-ni ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /^6178631/ /1/ ! voice translation-rule 2 rule 1 /^\(1...\)$/ /617863\1/ ! ! voice translation-profile in translate called 1 ! voice translation-profile out translate calling 2 ! ! voice-card 0 no dspfarm ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! class-map match-any CONTROL match ip dscp cs3 match ip dscp af31 class-map match-any RTP match ip dscp ef ! ! policy-map POLICY-CHECK class RTP priority percent 33 compress header ip rtp class CONTROL bandwidth percent 5 class class-default fair-queue ! ! ! ! ! interface Loopback0 ip address 10.10.110.2
Re: [OSL | CCIE_Voice] workbooks
i had several days of problems with files not opening, then working, then not, sections missing etc. Support did sort it out in the end, so as Wayne says that is the best route to take. I think the problem is on a case by case basis rather than generic. All working now - famous last words eh ! Regards Graham Hopkins On 21 Dec 2009, at 16:00, Wayne Lawson wrote: Have you guys contacted our support team?... Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 101 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com. On Dec 21, 2009, at 2:29 AM, Dew Swen dew.s...@gmail.com wrote: I'm in the same situation :/ - Dew Swen On Mon, Dec 21, 2009 at 8:10 AM, CCIE Downunder cciedownun...@gmail.com wrote: is it the error contacting server error or server's response could not be contacted. I think you have to download the damn workbook everytime you want to study or print the damn thing out. I agree with you its annoying. On Mon, Dec 21, 2009 at 5:52 AM, Robert McGhee bobwmcg...@verizon.net wrote: Is anyone else having issues opening the ipexpert workbooks? This is really getting annoying, seems like it happens every time I’m about to start a lab this happens…. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Sent from my MPhone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUE Email default-from address
Anyone know how to set the default from address when using email notification of a new voice mail.? Can't find it anywhere. Thanks Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE Email default-from address
Thanks, but I'm not sure this is what I need further question - this is an IOS
command rather than a Unity Express command - how is it linked to CUE and how
would it overwrite the default address that CUE uses ?
Regards
Graham Hopkins
On 16 Dec 2009, at 13:38, Rogers Ochieng wrote:
mta send mail-from
To specify a mail-from address (also called the RFC 821 envelope-from address
or the return-path address), use the mta send mail-from command in global
configuration mode. To remove this return-path information, use the no form
of this command.
mta send mail-from {hostname string | username string | username $s$}
no mta send mail-from {hostname string | username string | username $s$}
Syntax Description
hostname string
Simple Mail Transfer Protocol (SMTP) host name or IP address. If you specify
an IP address, you must enclose the IP address in brackets as follows:
[xxx.xxx.xxx.xxx].
username string
Sender username.
username $s$
Wildcard that specifies that the username is derived from the calling number.
-Original Message-
From: Graham Hopkins ghopk...@wolf-rock.co.uk
To: CCIE Voice Maillist ccie_voice@onlinestudylist.com
Date: Wed, 16 Dec 2009 12:34:08 +
Subject: [OSL | CCIE_Voice] CUE Email default-from address
Anyone know how to set the default from address when using email notification
of a new voice mail.? Can't find it anywhere.
Thanks
Graham Hopkins
___
For more information regarding industry leading CCIE Lab training, please
visit www.ipexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE Email default-from address
Yes, I tried that one it does alter the output from the
show fax configuration in CUE but didn't alter the emails sent for voicemail -
maybe its version dependant. Further you can set the text for email on per-user
basis in subscriber notification management but that doesn't appear in the
email either.
Regards
Graham Hopkins
On 16 Dec 2009, at 16:38, Rogers O. OCHIENG wrote:
I know I changed this sometime back on the web interface System Fax
Settings of the CUE web admin and it worked for voice mail notification too.
So I reckoned that it changes the mta settings as used in onramp fax config.
From: Graham Hopkins [mailto:ghopk...@wolf-rock.co.uk]
Sent: Wednesday, December 16, 2009 6:50 PM
To: Rogers Ochieng
Cc: CCIE Voice Maillist
Subject: Re: [OSL | CCIE_Voice] CUE Email default-from address
Thanks, but I'm not sure this is what I need further question - this is an
IOS command rather than a Unity Express command - how is it linked to CUE and
how would it overwrite the default address that CUE uses ?
Regards
Graham Hopkins
On 16 Dec 2009, at 13:38, Rogers Ochieng wrote:
mta send mail-from
To specify a mail-from address (also called the RFC 821 envelope-from address
or the return-path address), use the mta send mail-from command in global
configuration mode. To remove this return-path information, use the no form
of this command.
mta send mail-from {hostname string | username string | username $s$}
no mta send mail-from {hostname string | username string | username $s$}
Syntax Description
hostname string
Simple Mail Transfer Protocol (SMTP) host name or IP address. If you specify
an IP address, you must enclose the IP address in brackets as follows:
[xxx.xxx.xxx.xxx].
username string
Sender username.
username $s$
Wildcard that specifies that the username is derived from the calling number.
-Original Message-
From: Graham Hopkins ghopk...@wolf-rock.co.uk
To: CCIE Voice Maillist ccie_voice@onlinestudylist.com
Date: Wed, 16 Dec 2009 12:34:08 +
Subject: [OSL | CCIE_Voice] CUE Email default-from address
Anyone know how to set the default from address when using email notification
of a new voice mail.? Can't find it anywhere.
Thanks
Graham Hopkins
___
For more information regarding industry leading CCIE Lab training, please
visit www.ipexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Re: [OSL | CCIE_Voice] Members Area Access Restored and AdditionalFiles / Labs Available
Yes, most files I downloaded last week I could open, but the print option was no longer available. One gave a corrupt pdf message. Now I can't open any either. However the one file I have open is still good. Must remember not to reboot my laptop :-) Graham Hopkins On 13 Dec 2009, at 19:21, Adrian Clinton - Watkins wrote: I am getting the same issue. I have downloaded new content ok, having created a new account and does the transfer, but I cant open it with the same error your getting. Ive re-installed everything and tried mac and windows. I think the solution is to take the weekend off, relax spend time with the family and wait for guys at IPexpert who wont be doing that to get it fixed up. Adrian Clinton - Watkins CCIE #21806, CCDP, CCNP, CCVP, MCSE DDI: +44 (0)1905 825923 Office: +44 (0)1905 825900 Fax: +44 (0)1905 825901 Email: adrian.watk...@ggr.net Web: http://www.ggr.net For 24 hour technical support / assistance please call +44 (0) 1905 825 999 or email e-supp...@ggr.net GGR Communications Limited Company Registration Number: 2929785 De Salis House | De Salis Drive | Hampton Lovett Industrial Estate | Droitwich | Worcester | WR9 0QE -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIE Downunder Sent: 13 December 2009 6:03 PM To: Daryl Smith Cc: CCIE Voice Maillist; ccie...@onlinestudylist.com; ccie...@onlinestudylist.com; ccie_secur...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Members Area Access Restored and AdditionalFiles / Labs Available i have un-installed, re-installed adobe .. no success at all.. did the re-download and install work for you? On Mon, Dec 14, 2009 at 6:57 AM, Daryl Smith darylpsm...@gmail.com wrote: I have the same Issue I can't open my files I get Server Error. I've re-downloaded and re-installed the Adobe plug in No Dice. On 12/13/09 11:46 AM, CCIE Downunder cciedownun...@gmail.com wrote: i have to re-download. but i get this error when i try to open the downloaded file; There was an error opening this document. The file is damaged and could not be repaired Anyone else get this, and how do we fix it? this is getting frustrating, especially when you want to study On Fri, Dec 11, 2009 at 9:38 PM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: May also be worth pointing out that your old PDFs will now longer open with the new username/password and that you will need to re-download them - at least I did. Error was - Unable to get server data [FO error #2109, OS error #404] New files seem fine though. Regards Graham Hopkins On 10 Dec 2009, at 21:22, Ryan Barnum wrote: Members, Our Members Area account migration process has now been fully restored. You will have access to all of your files and some additional files that have been added for various workbooks. The process that should be completed is as follows: 1. Go to www.ipexpert.com click on ³Client Login² in the upper right corner. 2. You will fill in the ³New Customers² information which will include your first last name and email address (click on ³Sign up²). 3. You will need to confirm your email address, so check your email and follow the instructions pertaining to the confirmation procedures. 4. When you click on the link provided in your confirmation email, you will login w/ the credentials provided (email address and temporary password). 5. Once logged in, you will then be prompted to create your OWN password. 6. Once you have confirmed your email address and login you will see a blue box that says ³Migrate your old accounts here² click on the word ³here². 7. Enter your OLD USERNAME (which in many cases is different than your email address) and your OLD account password and then click ³Submit². 8. You will (should) then have access to all of your files under the ³Downloads² section. Note: You should use your new username / password for all logins (IPexpert Members Area, File Encryption, etc.) We apologize for the inconvenience, but are also happy to announce that the following additional labs have been added to your Members Area (if you have purchased the product): · CCIE RS Volume 2 is completed (will be added by Friday) · CCIE RS Volume 3 8 labs completed / available (will be added by Friday) · CCIE Security Volume 1 There¹s 1 lab that¹s not finished, all others are now available · CCIE Security Volume 2 There¹s 1 lab that¹s not finished, all others are now available
Re: [OSL | CCIE_Voice] Members Area Access Restored and Additional Files / Labs Available
May also be worth pointing out that your old PDFs will now longer open with the new username/password and that you will need to re-download them - at least I did. Error was - Unable to get server data [FO error #2109, OS error #404] New files seem fine though. Regards Graham Hopkins On 10 Dec 2009, at 21:22, Ryan Barnum wrote: Members, Our Members Area account migration process has now been fully restored. You will have access to all of your files – and some additional files that have been added for various workbooks. The process that should be completed is as follows: 1. Go to www.ipexpert.com – click on “Client Login” in the upper right corner. 2. You will fill in the “New Customers” information which will include your first last name and email address (click on “Sign up”). 3. You will need to confirm your email address, so check your email and follow the instructions pertaining to the confirmation procedures. 4. When you click on the link provided in your confirmation email, you will login w/ the credentials provided (email address and temporary password). 5. Once logged in, you will then be prompted to create your OWN password. 6. Once you have confirmed your email address and login – you will see a blue box that says “Migrate your old accounts here” – click on the word “here”. 7. Enter your OLD USERNAME (which in many cases is different than your email address) and your OLD account password and then click “Submit”. 8. You will (should) then have access to all of your files under the “Downloads” section. Note: You should use your new username / password for all logins (IPexpert Members Area, File Encryption, etc.) We apologize for the inconvenience, but are also happy to announce that the following additional labs have been added to your Members Area (if you have purchased the product): · CCIE RS Volume 2 is completed (will be added by Friday) · CCIE RS Volume 3 – 8 labs completed / available (will be added by Friday) · CCIE Security Volume 1 – There’s 1 lab that’s not finished, all others are now available · CCIE Security Volume 2 – There’s 1 lab that’s not finished, all others are now available · CCIE Voice Volume 1 was updated to include some fixes and additions (will be added by Friday) Please direct all questions to supp...@ipexpert.com – or you can reach them via LIVE CHAT. Regards, - Wayne Regards, Ryan Barnum Technical Support Engineer - IPexpert Mailto: rbar...@ipexpert.com Telephone: +1.810.326.1444, ext. 205 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] 7961
However I would be wary of the line on that site that reads Please note that this item is a license and not a boxed item Although how you have a refurbished license I don't know - best to double check - but then that's what CCIEs do isn't it :-) Graham On 10/09/2009 18:33, Jonathan Charles jonv...@gmail.com wrote: NM, they are refurbs... looks legit Ingram Micro has them for just a tad less... Jonathan On Thu, Sep 10, 2009 at 12:31 PM, Jonathan Charles jonv...@gmail.com wrote: That looks really gray to me... J On Thu, Sep 10, 2009 at 10:51 AM, Mark Holloway m...@markholloway.com wrote: http://www.costcentral.com/proddetail/Cisco_IP_Phone_7961G/CP7961GRF/1036946 7/ These guys seem to have great prices for any 7961/62/65 model of phone. On Sep 10, 2009, at 8:36 AM, Kevin Damisch wrote: http://letmegooglethatforyou.com/?q=used+cisco+7961 Just have to go shopping or try ebay. Thanks, Kevin -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com ] On Behalf Of Mauricio Aduna Sent: Thursday, September 10, 2009 10:29 AM To: m...@markholloway.com Cc: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] 7961 Hi Mark, I was wondering if you could be so kind to share with us where do you find the 7961 phones for $140? Thank you! Best Regards, Maurici Aduna ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUCME - SIP Phone Issues LAB3A
In LAB 3A the SIP Phones do not display the date and time ( my phones are 7960Gs) The NTP server is set as voice register global ntp-server 10.10.210.1 mode unicast but the SIPDefault.cnf file only contains sntp_mode: directedbroadcast; sntp_server: 0.0.0.0; the phone status shows parse errors in both the SIPDefault.cnf and the phone specific file, also shows a E102 no time server error. Also the no conference enable on the template doesn't seem to work either, there are some .xml softkey files but they all have Confrn in them Current phone load is P0S3=08-6-00, but I have tried other versions, CUCME is 7.1 on 12.4(24)T1 Any ideas? Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX 7 Web Browser Issues
Thanks Vik, but that has the same problem. Going to rebuild the server (its on vmware) with all the latest patches. Looking forward to the Linux Appliance version ! On 03/07/2009 19:36, Vik Malhi vma...@ipexpert.com wrote: Use Remote Desktop (MSTSC) to get on the UCCX and use browser on the server.
Re: [OSL | CCIE_Voice] UCCX 7 Web Browser Issues
Well rebuild fixed this, Ok with Remote Desktop and other browsers, must have corrupted something on the previous build Graham Hopkins On 04/07/2009 09:40, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: Thanks Vik, but that has the same problem. Going to rebuild the server (its on vmware) with all the latest patches. Looking forward to the Linux Appliance version ! On 03/07/2009 19:36, Vik Malhi vma...@ipexpert.com wrote: Use Remote Desktop (MSTSC) to get on the UCCX and use browser on the server.
[OSL | CCIE_Voice] UCCX 7 Web Browser Issues
Having a problem with UCCX 7 in that the drop down menus don¹t work in any browser I have tested, including the version of IE installed on the server. Probably a javascript type issue but I cannot pin it down any ideas please ? Graham
