Re: [OSL | CCIE_Voice] Media Resources Backup Redundancy
I think the problem is that if you are using a transcoding resource at BR1 then your stream across the WAN would be G711. In the scenario you give only 729 is allowed on the WAN. Example given: Scenario 1 MoH Server -G711-HQ Xcoder-WAN-G729-BR1 Phone Scenario 2 MoH Server -G711-WAN-BR1 Xcoder-G729-BR1 Phone I think CFB resources as backups would work as long as there are transcoding resources available. Scenario 1 HQ CFB -G711-HQ Xcoder-WAN-G729-BR1 Phone Scenario 2 HQ Phone -WAN-G729-BR1 CFB From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Devildoc Sent: Thursday, August 28, 2008 10:12 AM To: CCIE Voice Online Study List Subject: [OSL | CCIE_Voice] Media Resources Backup Redundancy Hello, Would it be possible for one location to use media resources (confbridge and transcoder) from another remote location? For example, can HQ users use media resources located at the BR1 location? I know it's possible because many exercises in the workbook ask you configure media resources redundancy. However, when i configured it, it never worked. As an example, I tried to configure MOH unicast stream from HQ to BR1 over the WAN link. The WAN link only allows G729 codec, but I configured the MOH unicast to support only G711 codec. I also configured the MOH server to have a transcoder resource located in the BR1 location. To test it, I placed a call from HQ to BR1 and placed the call on hold. I then checked the performance counter and it turned out that the BR1 transcoder never engaged, so the MOH unicast was never played. If I swapped out the BR1 transcoder for the HQ transcoder, then the that worked fine. So if this doesn't work, then how would you configure media resources for backup redundancy? Thanks. JD Be the filmmaker you always wanted to be-learn how to burn a DVD with Windows(r). Make your smash hit http://clk.atdmt.com/MRT/go/108588797/direct/01/
Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap
Voice translation-rule 1 Rule 1 /^1\($\)/ /\1/ (I would prefer this if allowed: Rule 1 /^1\(4...$\)/ /\1/) ! Voice Translation-Profile voicemail Translate redirect-called 1 ! Dial-peer 2000 voip Destination-pattern 1$ ( It would be better to do 14...$ if that's allowed) Translation-profile incoming voicemail Session protocol sipv2 Session target ipv4:CUE IP Address Ip qos dscp cs3 signaling Fax rate disable Dtmf-relay sip-notify Codec g711ulaw No vad ! Dial-peer 3000 voip Destination-pattern [12]...T (It would be better to make the patter [12]00. Or something like that to avoid the need to wait for an interdigit timeout) Session target ras Dtmf-relay alpha-numeric Tech-prefix 1 Ip qos dscp cs3 signaling ! telephony-service timeouts interdigit 3 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Arevalo Sent: Wednesday, August 27, 2008 8:19 AM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap Hi, I have seen/done several exercises about access directly to a users's mailbox by dialing a digit, like 4 or 8, but, what if you are requested to dial a digit that overlaps current dial plan? For example: dial number 1 then any other 4 digits, that would be 1, but the HQ's DNs are 1 plus three any digits, ie: 1XXX. How could we do the same in the unity, so the access to HQ DNs beginning with 1 dont be affected, remember it should be already a dial-peer with destination-pattern [12]... pointing to ras. I have been trying to find a solution but no luck so far. Slds//r.a.
Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap
CUCM Unity should work using the RP/VM Profile method you will just have to wait for the interdigit timeout. In the real world I would use an * or # as a steering digit rather than a number. From: Ricardo Arevalo [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 27, 2008 10:08 AM To: Rimon Vallavanatt Jr. Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap I will try with that example, besides i have tested similar dial peers, but the closest match always take precedence. I'll check this out and let you know. And what about the call routing in CCM - Unity? Have any ideas? //r.a. On Wed, Aug 27, 2008 at 9:38 AM, Rimon Vallavanatt Jr. [EMAIL PROTECTED] wrote: Voice translation-rule 1 Rule 1 /^1\($\)/ /\1/ (I would prefer this if allowed: Rule 1 /^1\(4...$\)/ /\1/) ! Voice Translation-Profile voicemail Translate redirect-called 1 ! Dial-peer 2000 voip Destination-pattern 1$ ( It would be better to do 14...$ if that's allowed) Translation-profile incoming voicemail Session protocol sipv2 Session target ipv4:CUE IP Address Ip qos dscp cs3 signaling Fax rate disable Dtmf-relay sip-notify Codec g711ulaw No vad ! Dial-peer 3000 voip Destination-pattern [12]...T (It would be better to make the patter [12]00. Or something like that to avoid the need to wait for an interdigit timeout) Session target ras Dtmf-relay alpha-numeric Tech-prefix 1 Ip qos dscp cs3 signaling ! telephony-service timeouts interdigit 3 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Arevalo Sent: Wednesday, August 27, 2008 8:19 AM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap Hi, I have seen/done several exercises about access directly to a users's mailbox by dialing a digit, like 4 or 8, but, what if you are requested to dial a digit that overlaps current dial plan? For example: dial number 1 then any other 4 digits, that would be 1, but the HQ's DNs are 1 plus three any digits, ie: 1XXX. How could we do the same in the unity, so the access to HQ DNs beginning with 1 dont be affected, remember it should be already a dial-peer with destination-pattern [12]... pointing to ras. I have been trying to find a solution but no luck so far. Slds//r.a.
Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap
Assuming you don't have any phones with DN's in the 1[12]XX range, you could make your RP 1[12]XXX then you won't have the interdigit timeout issue. From: Ricardo Arevalo [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 27, 2008 10:38 AM To: Rimon Vallavanatt Jr. Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap Yes i did it with a RP (1) forwarded to VM, but the interdigit timeout now affects the extension-to-extension calls, the call forward to VM by dialing 1 works as expected, the problem is that now every internal call to any DN whitin range 1XXX has to wait the interdigit timeout. I would like to know if there is any way to activate something like urgent priority used in route-patterns and active by default in translation patterns, but applied to match internal DN numbers, i haven't found it so far. In the real world its a very bad practice to use an existing DN leading digit as the steering digit to any feature, i have used * as well. //r.a. On Wed, Aug 27, 2008 at 11:12 AM, Rimon Vallavanatt Jr. [EMAIL PROTECTED] wrote: CUCM Unity should work using the RP/VM Profile method you will just have to wait for the interdigit timeout. In the real world I would use an * or # as a steering digit rather than a number. From: Ricardo Arevalo [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 27, 2008 10:08 AM To: Rimon Vallavanatt Jr. Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap I will try with that example, besides i have tested similar dial peers, but the closest match always take precedence. I'll check this out and let you know. And what about the call routing in CCM - Unity? Have any ideas? //r.a. On Wed, Aug 27, 2008 at 9:38 AM, Rimon Vallavanatt Jr. [EMAIL PROTECTED] wrote: Voice translation-rule 1 Rule 1 /^1\($\)/ /\1/ (I would prefer this if allowed: Rule 1 /^1\(4...$\)/ /\1/) ! Voice Translation-Profile voicemail Translate redirect-called 1 ! Dial-peer 2000 voip Destination-pattern 1$ ( It would be better to do 14...$ if that's allowed) Translation-profile incoming voicemail Session protocol sipv2 Session target ipv4:CUE IP Address Ip qos dscp cs3 signaling Fax rate disable Dtmf-relay sip-notify Codec g711ulaw No vad ! Dial-peer 3000 voip Destination-pattern [12]...T (It would be better to make the patter [12]00. Or something like that to avoid the need to wait for an interdigit timeout) Session target ras Dtmf-relay alpha-numeric Tech-prefix 1 Ip qos dscp cs3 signaling ! telephony-service timeouts interdigit 3 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Arevalo Sent: Wednesday, August 27, 2008 8:19 AM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap Hi, I have seen/done several exercises about access directly to a users's mailbox by dialing a digit, like 4 or 8, but, what if you are requested to dial a digit that overlaps current dial plan? For example: dial number 1 then any other 4 digits, that would be 1, but the HQ's DNs are 1 plus three any digits, ie: 1XXX. How could we do the same in the unity, so the access to HQ DNs beginning with 1 dont be affected, remember it should be already a dial-peer with destination-pattern [12]... pointing to ras. I have been trying to find a solution but no luck so far. Slds//r.a.
Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap
I think any kind of urgent priority patterns would prevent you from dialing 2 numbers and would not meet the requirement. 2001 - urgent Can't dial 20012 (routes at 2001) You could decrease the interdigit timer on CUCM to 3 secs so that you don't have to wait as long. That's what I was doing in CME. From: Ricardo Arevalo [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 27, 2008 11:10 AM To: Rimon Vallavanatt Jr. Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap I know, that would be the easy way... unfortunately that is not the request. The request is to dial 1 plus four digits from 0 to 9 in both CCM to Unity VM and CME to CUE, ie: 1 //r.a. On Wed, Aug 27, 2008 at 11:47 AM, Rimon Vallavanatt Jr. [EMAIL PROTECTED] wrote: Assuming you don't have any phones with DN's in the 1[12]XX range, you could make your RP 1[12]XXX then you won't have the interdigit timeout issue. From: Ricardo Arevalo [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 27, 2008 10:38 AM To: Rimon Vallavanatt Jr. Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap Yes i did it with a RP (1) forwarded to VM, but the interdigit timeout now affects the extension-to-extension calls, the call forward to VM by dialing 1 works as expected, the problem is that now every internal call to any DN whitin range 1XXX has to wait the interdigit timeout. I would like to know if there is any way to activate something like urgent priority used in route-patterns and active by default in translation patterns, but applied to match internal DN numbers, i haven't found it so far. In the real world its a very bad practice to use an existing DN leading digit as the steering digit to any feature, i have used * as well. //r.a. On Wed, Aug 27, 2008 at 11:12 AM, Rimon Vallavanatt Jr. [EMAIL PROTECTED] wrote: CUCM Unity should work using the RP/VM Profile method you will just have to wait for the interdigit timeout. In the real world I would use an * or # as a steering digit rather than a number. From: Ricardo Arevalo [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 27, 2008 10:08 AM To: Rimon Vallavanatt Jr. Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap I will try with that example, besides i have tested similar dial peers, but the closest match always take precedence. I'll check this out and let you know. And what about the call routing in CCM - Unity? Have any ideas? //r.a. On Wed, Aug 27, 2008 at 9:38 AM, Rimon Vallavanatt Jr. [EMAIL PROTECTED] wrote: Voice translation-rule 1 Rule 1 /^1\($\)/ /\1/ (I would prefer this if allowed: Rule 1 /^1\(4...$\)/ /\1/) ! Voice Translation-Profile voicemail Translate redirect-called 1 ! Dial-peer 2000 voip Destination-pattern 1$ ( It would be better to do 14...$ if that's allowed) Translation-profile incoming voicemail Session protocol sipv2 Session target ipv4:CUE IP Address Ip qos dscp cs3 signaling Fax rate disable Dtmf-relay sip-notify Codec g711ulaw No vad ! Dial-peer 3000 voip Destination-pattern [12]...T (It would be better to make the patter [12]00. Or something like that to avoid the need to wait for an interdigit timeout) Session target ras Dtmf-relay alpha-numeric Tech-prefix 1 Ip qos dscp cs3 signaling ! telephony-service timeouts interdigit 3 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Arevalo Sent: Wednesday, August 27, 2008 8:19 AM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Direct Access to unity/cue with overlap Hi, I have seen/done several exercises about access directly to a users's mailbox by dialing a digit, like 4 or 8, but, what if you are requested to dial a digit that overlaps current dial plan? For example: dial number 1 then any other 4 digits, that would be 1, but the HQ's DNs are 1 plus three any digits, ie: 1XXX. How could we do the same in the unity, so the access to HQ DNs beginning with 1 dont be affected, remember it should be already a dial-peer with destination-pattern [12]... pointing to ras. I have been trying to find a solution but no luck so far. Slds//r.a.
Re: [OSL | CCIE_Voice] Search at Mailling list arquive
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sérgio Polizer Sent: Tuesday, August 12, 2008 6:31 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Search at Mailling list arquive Hi, Is there a way to make a find at the mailing list archive? I tried with Google w/ something like: key word site:www.onlinestudylist, but did not work. Thanks in advance, Sergio Polizer. Conheça já o Windows Live Spaces, o site de relacionamentos do Messenger! Crie já o seu! http://www.amigosdomessenger.com.br
Re: [OSL | CCIE_Voice] Translation Rule for ephone-dn
Try incoming instead of outgoing on ephone ephone-dn 1 number 8001 translation-profile incoming HELP -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WorkerBee Sent: Wednesday, June 18, 2008 4:06 PM To: OSL CCIE Voice Lab Exam; [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Translation Rule for ephone-dn Numexp will work but I want to use the newer voice translation-profile method applied to ephone-dn instead. Under ephone-dn, it does support translation-profile but it doesn't seems to work. On Thu, Jun 19, 2008 at 5:02 AM, Derrick Shumake [EMAIL PROTECTED] wrote: Try using numexp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WorkerBee Sent: Wednesday, June 18, 2008 1:58 PM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Translation Rule for ephone-dn I would like to perform digit manipulation at ephone-dn level. When a user using phone DN 8001 call HELP-4357, it will automatically translate to 028. I tried on voice translation-profile, it does not work but it works if I use the older translate command. == not working == voice translation-rule 2 rule 1 /4357/ /028/ voice translation-profile HELP translate called 2 ephone-dn 1 number 8001 translation-profile outgoing HELP == working == translation-rule 2 Rule 0 4357 028 ephone-dn 1 number 8001 translate called 2 Any idea? Thanks.
[OSL | CCIE_Voice] Gatekeeper 1st call g711 2nd call g729 - works but not as expected
I have setup the following: Configured two regions on the CCM, one that talks G.711 to everything else and one that talks G.729 to everything else. Created two DP, GK-711 and GK-729 with their respective regions. I registered the GK in call manager and then created two trunks. One using the GK-G711-DP and another using GK-G729-DP. Then created one route group with both trunks using top down distribution with GK-711-Trunk first and GK-G729-Trunk second. Created a RL and RP to point to the Route Group. I set BRQ to true on the CUCMs. I've also tried it with two RGs. I've tried it with the voice class codec and with two different dial-peers, one with 711, one with 729. It works just fine. The gatekeeper shows one call 711 one call 729. The phones on the CME , if I hit the ? button show what I would expect. The problem is that at the HQ site the phones both show 711 when I hit the ? button. I verified that the 729 stream is being transcoded to g711. My question is why? Thanks, Rimon Vallavanatt Jr. Director, Installations Phone:713.881.7133 Fax:713.881.7233 image001.png
Re: [OSL | CCIE_Voice] Gatekeeper 1st call g711 2nd call g729 - worksbut not as expected
I removed the transcoding resources from my MRGLs and now the phones at HQ work as expected. Why is transcoding being invoked when transcoding resources are present?
Re: [OSL | CCIE_Voice] Default DSCP Marking on CME IP Phone
So if there is a switch at the CME site do we need to configure QoS to match af31 or rewrite the signaling DSCP values from the phones? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Devildoc Sent: Wednesday, May 07, 2008 12:48 PM To: Christian Narvaez; CCIE Voice Online Study List Subject: Re: [OSL | CCIE_Voice] Default DSCP Marking on CME IP Phone Thanks Christian, I was hoping for some configuration at the telephony level or the phone level but i couldn't find anything like that. Oh well, thanks for the info. JD Subject: RE: [OSL | CCIE_Voice] Default DSCP Marking on CME IP Phone Date: Tue, 6 May 2008 18:23:16 -0400 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com I think you should configure ip qos dscp 24 signaling under the dial-peers voice of the CME -Original Message- From: [EMAIL PROTECTED] on behalf of Devildoc Sent: Tue 5/6/2008 12:29 PM To: CCIE Voice Online Study List Subject: [OSL | CCIE_Voice] Default DSCP Marking on CME IP Phone Hello, The default DSCP value of the call control for IP Phones registered to CCM is CS3 and AF31 for CME. Does anyone know a way to change the default DSCP value of the call control for IP Phones registered to CME to a value of CS3? Thanks for any input. JD _ With Windows Live for mobile, your contacts travel with you. http://www.windowslive.com/mobile/overview.html?ocid=TXT_TAGLM_WL_Refres h_mobile_052008 Make Windows Vista more reliable and secure with Windows Vista Service Pack 1. Learn more. http://www.windowsvista.com/SP1?WT.mc_id=hotmailvistasp1banner
[OSL | CCIE_Voice] AAR, TEHO, backup gws
I'm confused as to how AAR should work when you factor in backup gateways. It's easy enough to exclude TEHO patterns from the AAR CSS but when local, ld, or intl patterns have backup gateways it seems to me that AAR should not use them either. If that is the case then wouldn't it be better to just create a separate dial-plan for AAR? Since that would be very time consuming what do you think about just creating a single pattern for AAR and giving each site's AAR CSS access to only that? e.g. css-hq-aar includes only pt-hq-aar and pattern 9.@ in pt-hq-aar that points to a RL with only the local gw. css-br1-aar includes only pt-br1-aar and pattern 9.@ in pt-br1-aar that points to a RL with only the local gw. Rimon Vallavanatt Jr. Director, Installations Phone:713.881.7133 Fax:713.881.7233 image001.png
