[OSL | CCIE_Voice] Lab 1 - MWI
Hello, I am working on volume 2 lab 1, when hq phone calls br2 phone and hq phone leave a message on br2 phone, it gets to the voicemail but the MWI does not turn. Any one with an idea of what could be the problem. Below is the relevant config dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay rtp-nte codec g711ulaw no vad ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 2 sdspfarm tag 1 br2-xcoder no auto-reg-ephone authentication credential admin cisco max-ephones 10 max-dn 10 ip source-address 10.10.110.3 port 2000 url services http://10.10.202.2/voiceview/common/login.do url authentication http://10.10.202.1/CCMCIP/authenticate.asp voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T web admin system name admin password cisco transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Apr 08 2010 23:30:55 ephone-dn 1 dual-line number 3001 no-reg primary label Br2 Phn1 name Br2 Phn1 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 2 dual-line number 3002 no-reg primary label Br2 Phn2 name Br2 Phn2 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 3 number 3999 no-reg primary mwi on ! ! ephone-dn 4 number 3998 no-reg primary mwi off ! ! ephone 1 device-security-mode none mac-address 001E.0B2D.F37D username Br2Phn1 password cisco type CIPC button 1:1 ! ! ! ephone 2 device-security-mode none mac-address 001E.EC15.996D username Br2Phn2 password cisco type CIPC button 1:2 ! thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on HQ3750 SW...can not browse into CUCM
Alex: Just a question, can you see the cucm interface as full duplex with show cdp neibor det from hq sw? I manage to solve this issue ( I don't get the cdp duplex mismatch message), but I see the interface as half with show cdp neibor det command Thanks Subject: RE: [OSL | CCIE_Voice] %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on HQ3750 SW...can not browse into CUCM Date: Thu, 8 Apr 2010 10:36:20 -0700 From: alcol...@cisco.com To: gorr...@hotmail.com; ar...@ipexpert.com; siddas...@gmail.com; mthompson...@gmail.com CC: ccie_voice@onlinestudylist.com If someone can get a message to Proctorlabs it could save a lot of recurring posts about this issue... I was able to correct this duplex error on my lab ESXi servers without rebuilding. These errors were driving me crazy. After reinstalling the virtual NIC in vmware, the problems went away completely. See the blog post: I experienced this duplex mismatch problem using ESXi and cucm 7.0.2. The problem only happened with the Pub showing up as half duplex at the switch, no matter what I set it to. The Pub cli show network eth0 showed 100/full no matter what. Finally, I shut the CUCM down. Went to the VMware settings for the Pub and deleted the existing NIC. I reinstalled an E1000 NIC. When I restarted the Pub, it was connected at 1000/full, which is what I wanted. It's possible reinstalling the flex NIC type may also work. Message was edited by: shockinac From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Angel Perez Sent: Thursday, April 08, 2010 12:28 PM To: ar...@ipexpert.com; siddas...@gmail.com; mthompson...@gmail.com Cc: osl osl Subject: Re: [OSL | CCIE_Voice] %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on HQ3750 SW...can not browse into CUCM I agree with you Amy, setting the duplex to half is not a good idea Date: Thu, 8 Apr 2010 12:57:28 -0400 From: ar...@ipexpert.com To: siddas...@gmail.com; mthompson...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on HQ3750 SW...can not browse into CUCM This message will not interfere with access to the servers. It is a “cosmetic issue” with vmware. It is best not to alter the duplex settings on the switchport and just disable cdp for that port only, as previously stated. Int fa1/0/4 no cdp enable If you are having issues browsing to CUCM, first verify you can ping CUCM from HQ-RTR. If that is successful, try to ping from your computer. If that fails, I would advise checking your VPN connectivity to Proctor Labs. HTH, Amy --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 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, 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: Ashar Siddiqui siddas...@gmail.com Date: Thu, 08 Apr 2010 16:22:01 +0100 To: Mike Thompson mthompson...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on HQ3750 SW...can not browse into CUCM Guys, There is nothing to worry about this, it happens to me all the time at my home lab as well. I think this has something to do with VMware. At Ipexert rack, I just use to change the duplex to half at the start and everything works fineeven if you don't change it, it will still work but those messages will pop up at switch every after few seconds which is kind of nuisance. Ash n 08/04/2010 06:19, Mike Thompson wrote: Utils service list will show the list of services and their status From: Ryan Schwab [mailto:schwab...@shaw.ca] Sent: Thursday, April 08, 2010 1:15 AM To: 'Mike Thompson'; 'vccie2010'; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on HQ3750 SW...can not browse into CUCM Mike, on that note, are there any specific CLI commands you can think of to verify this? :) From: Mike Thompson [mailto:mthompson...@gmail.com] Sent: April-07-10 11:16 PM To: 'Ryan Schwab'; 'vccie2010'; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on HQ3750 SW...can not browse into CUCM I would agree, first things is to verify that Tomcat is happy. From: ccie_voice-boun...@onlinestudylist.com
[OSL | CCIE_Voice] CME in SRST dial-peer issue
Hi all: I've the following issue with cme as srst: When I've configured telephony-service srst mode auto-provision all srst dn line-mode octo max-ephones 2 max-dn 20 preference 2 no-reg then i shut down the serial interface and the phones register to srts router, then I no shut the ser interface and the phones register back to cucm. At this point i do: sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000 voip up up 3...$0syst ipv4:140.50.64.20 3001 voip up up 3...$1syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up down2 50/0/1 20002 pots up down2 50/0/2 The ephone-dns dial-peer created are down and with no des pattern. After some changes to telephony-ser configuration when the phones go to srst and then back to cucm I have the following result: #sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000voip up up 3...$ 0 syst ipv4:140.50.64.20 3001voip up up 3...$ 1 syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up up 3002$ 2 50/0/1 20002 pots up up 3001$ 2 50/0/2 So when i try to call ephone 3001 let say from pstn the call fails becouse dial-peer 20002 is a closer match than 3...$ (dial peer 3001 poiting to CUCM), I know that I can chage this behaviour with dial-peer hunt ? 0-7 Dial-peer hunting choices, listed in hunting order within each choice: 0 - Longest match in phone number, explicit preference, random selection. 1 - Longest match in phone number, explicit preference, least recent use. 2 - Explicit preference, longest match in phone number, random selection. 3 - Explicit preference, longest match in phone number, least recent use. 4 - Least recent use, longest match in phone number, explicit preference. 5 - Least recent use, explicit preference, longest match in phone number. 6 - Random selection. 7 - Least recent use. dial-peer hunt 3 This way explicit preference will be checked before, but my question is: What changes in cme as srst make dial-peer to be persistent when the gw is NOT in srst? I think that this behaviour is the result of create cnf command but I'm not sure Thanks in advance _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 - MWI
Hi, Have you checked with debub voip dialpeer that the CUE dials your MWI on/off numbers? There is a bug that sometimes makes it use the default MWI extensions. I believe that they are and 8889 If so change the MWI settings temporary in CUE to not include outdial, then do a resync of MWI and look at the debug, you should not get any output. Then change it back to outdial and resync once more, this time you should get output in the debug. Check the debug to see that CUE now uses your MWI numbers. Roger Källberg Unified Communication Consultant Cygate AB From: Omotayo [mailto:adefilabi...@gmail.com] Sent: den 9 april 2010 09:01 To: OSL Group Subject: [OSL | CCIE_Voice] Lab 1 - MWI Hello, I am working on volume 2 lab 1, when hq phone calls br2 phone and hq phone leave a message on br2 phone, it gets to the voicemail but the MWI does not turn. Any one with an idea of what could be the problem. Below is the relevant config dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay rtp-nte codec g711ulaw no vad ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 2 sdspfarm tag 1 br2-xcoder no auto-reg-ephone authentication credential admin cisco max-ephones 10 max-dn 10 ip source-address 10.10.110.3 port 2000 url services http://10.10.202.2/voiceview/common/login.do url authentication http://10.10.202.1/CCMCIP/authenticate.asp voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T web admin system name admin password cisco transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Apr 08 2010 23:30:55 ephone-dn 1 dual-line number 3001 no-reg primary label Br2 Phn1 name Br2 Phn1 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 2 dual-line number 3002 no-reg primary label Br2 Phn2 name Br2 Phn2 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 3 number 3999 no-reg primary mwi on ! ! ephone-dn 4 number 3998 no-reg primary mwi off ! ! ephone 1 device-security-mode none mac-address 001E.0B2D.F37D username Br2Phn1 password cisco type CIPC button 1:1 ! ! ! ephone 2 device-security-mode none mac-address 001E.EC15.996D username Br2Phn2 password cisco type CIPC button 1:2 ! thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME in SRST dial-peer issue
Hi, If I'm not completely wrong I do believe that Vik mentioned this at the ILT in March. As I can recall it's a bug in the CME version that we have in the lab or possibly it's IOS related. Maybe Vik can verify/clarify this? Roger Källberg Unified Communication Consultant Cygate AB From: Angel Perez [mailto:gorr...@hotmail.com] Sent: den 9 april 2010 09:53 To: osl osl Subject: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi all: I've the following issue with cme as srst: When I've configured telephony-service srst mode auto-provision all srst dn line-mode octo max-ephones 2 max-dn 20 preference 2 no-reg then i shut down the serial interface and the phones register to srts router, then I no shut the ser interface and the phones register back to cucm. At this point i do: sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000 voip up up 3...$0syst ipv4:140.50.64.20 3001 voip up up 3...$1syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up down2 50/0/1 20002 pots up down2 50/0/2 The ephone-dns dial-peer created are down and with no des pattern. After some changes to telephony-ser configuration when the phones go to srst and then back to cucm I have the following result: #sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000voip up up 3...$ 0 syst ipv4:140.50.64.20 3001voip up up 3...$ 1 syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up up 3002$ 2 50/0/1 20002 pots up up 3001$ 2 50/0/2 So when i try to call ephone 3001 let say from pstn the call fails becouse dial-peer 20002 is a closer match than 3...$ (dial peer 3001 poiting to CUCM), I know that I can chage this behaviour with dial-peer hunt ? 0-7 Dial-peer hunting choices, listed in hunting order within each choice: 0 - Longest match in phone number, explicit preference, random selection. 1 - Longest match in phone number, explicit preference, least recent use. 2 - Explicit preference, longest match in phone number, random selection. 3 - Explicit preference, longest match in phone number, least recent use. 4 - Least recent use, longest match in phone number, explicit preference. 5 - Least recent use, explicit preference, longest match in phone number. 6 - Random selection. 7 - Least recent use. dial-peer hunt 3 This way explicit preference will be checked before, but my question is: What changes in cme as srst make dial-peer to be persistent when the gw is NOT in srst? I think that this behaviour is the result of create cnf command but I'm not sure Thanks in advance Hotmail: Free, trusted and rich email service. Get it now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 - MWI
Hello, the CUE uses the configured MWI -outdialling I will check the debug to see if its been used Regards 2010/4/9 Roger Källberg roger.kallb...@cygate.se Hi, Have you checked with debub voip dialpeer that the CUE dials your MWI on/off numbers? There is a bug that sometimes makes it use the default MWI extensions. I believe that they are …. and 8889…. If so change the MWI settings temporary in CUE to not include outdial, then do a resync of MWI and look at the debug, you should not get any output. Then change it back to outdial and resync once more, this time you should get output in the debug. Check the debug to see that CUE now uses your MWI numbers. *Roger Källberg* Unified Communication Consultant Cygate AB *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* den 9 april 2010 09:01 *To:* OSL Group *Subject:* [OSL | CCIE_Voice] Lab 1 - MWI Hello, I am working on volume 2 lab 1, when hq phone calls br2 phone and hq phone leave a message on br2 phone, it gets to the voicemail but the MWI does not turn. Any one with an idea of what could be the problem. Below is the relevant config dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay rtp-nte codec g711ulaw no vad ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 2 sdspfarm tag 1 br2-xcoder no auto-reg-ephone authentication credential admin cisco max-ephones 10 max-dn 10 ip source-address 10.10.110.3 port 2000 url services http://10.10.202.2/voiceview/common/login.do url authentication http://10.10.202.1/CCMCIP/authenticate.asp voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T web admin system name admin password cisco transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Apr 08 2010 23:30:55 ephone-dn 1 dual-line number 3001 no-reg primary label Br2 Phn1 name Br2 Phn1 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 2 dual-line number 3002 no-reg primary label Br2 Phn2 name Br2 Phn2 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 3 number 3999 no-reg primary mwi on ! ! ephone-dn 4 number 3998 no-reg primary mwi off ! ! ephone 1 device-security-mode none mac-address 001E.0B2D.F37D username Br2Phn1 password cisco type CIPC button 1:1 ! ! ! ephone 2 device-security-mode none mac-address 001E.EC15.996D username Br2Phn2 password cisco type CIPC button 1:2 ! thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME in SRST dial-peer issue
Hi: Yes it could be a bug, becouse from my experience the behaviour is not uniform with this issue Also when I change ephone name or label somentimes the ephone doesn't get a dn and it register without dn... another strange issue too I'm running IOS 12.4(20)T Thanks From: roger.kallb...@cygate.se To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com CC: vma...@ipexpert.com Date: Fri, 9 Apr 2010 10:32:12 +0200 Subject: RE: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi, If I’m not completely wrong I do believe that Vik mentioned this at the ILT in March. As I can recall it’s a bug in the CME version that we have in the lab or possibly it’s IOS related. Maybe Vik can verify/clarify this? Roger Källberg Unified Communication Consultant Cygate AB From: Angel Perez [mailto:gorr...@hotmail.com] Sent: den 9 april 2010 09:53 To: osl osl Subject: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi all: I've the following issue with cme as srst: When I've configured telephony-service srst mode auto-provision all srst dn line-mode octo max-ephones 2 max-dn 20 preference 2 no-reg then i shut down the serial interface and the phones register to srts router, then I no shut the ser interface and the phones register back to cucm. At this point i do: sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000 voip up up 3...$0syst ipv4:140.50.64.20 3001 voip up up 3...$1syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up down2 50/0/1 20002 pots up down2 50/0/2 The ephone-dns dial-peer created are down and with no des pattern. After some changes to telephony-ser configuration when the phones go to srst and then back to cucm I have the following result: #sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000voip up up 3...$ 0 syst ipv4:140.50.64.20 3001voip up up 3...$ 1 syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up up 3002$ 2 50/0/1 20002 pots up up 3001$ 2 50/0/2 So when i try to call ephone 3001 let say from pstn the call fails becouse dial-peer 20002 is a closer match than 3...$ (dial peer 3001 poiting to CUCM), I know that I can chage this behaviour with dial-peer hunt ? 0-7 Dial-peer hunting choices, listed in hunting order within each choice: 0 - Longest match in phone number, explicit preference, random selection. 1 - Longest match in phone number, explicit preference, least recent use. 2 - Explicit preference, longest match in phone number, random selection. 3 - Explicit preference, longest match in phone number, least recent use. 4 - Least recent use, longest match in phone number, explicit preference. 5 - Least recent use, explicit preference, longest match in phone number. 6 - Random selection. 7 - Least recent use. dial-peer hunt 3 This way explicit preference will be checked before, but my question is: What changes in cme as srst make dial-peer to be persistent when the gw is NOT in srst? I think that this behaviour is the result of create cnf command but I'm not sure Thanks in advance Hotmail: Free, trusted and rich email service. Get it now. _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 13A : CUPC Deskphone mode not working and CUPC shows Connection(Limited)
Hello, Please refer to the following blog entry, http://blog.ipexpert.com/cti-phone-control-from-cupc/#more-2272 hth, On Thu, Apr 8, 2010 at 11:16 PM, vccie2010 vccie2...@gmail.com wrote: I am doing Lab 13A : CUPC Deskphone mode not working and CUPC shows Connection(Limited). I have followed the steps in verbatim. I am labbing it on IPX remote laba nd have CUPC on my laptop. I use SW VPN. Am I missing something here or is it becoz of VPN. thanks for your help... -M ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME in SRST dial-peer issue
Sorry I meant 12.4(20)T2 From: gorr...@hotmail.com To: roger.kallb...@cygate.se; ccie_voice@onlinestudylist.com Date: Fri, 9 Apr 2010 09:47:54 + CC: vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi: Yes it could be a bug, becouse from my experience the behaviour is not uniform with this issue Also when I change ephone name or label somentimes the ephone doesn't get a dn and it register without dn... another strange issue too I'm running IOS 12.4(20)T Thanks From: roger.kallb...@cygate.se To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com CC: vma...@ipexpert.com Date: Fri, 9 Apr 2010 10:32:12 +0200 Subject: RE: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi, If I’m not completely wrong I do believe that Vik mentioned this at the ILT in March. As I can recall it’s a bug in the CME version that we have in the lab or possibly it’s IOS related. Maybe Vik can verify/clarify this? Roger Källberg Unified Communication Consultant Cygate AB From: Angel Perez [mailto:gorr...@hotmail.com] Sent: den 9 april 2010 09:53 To: osl osl Subject: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi all: I've the following issue with cme as srst: When I've configured telephony-service srst mode auto-provision all srst dn line-mode octo max-ephones 2 max-dn 20 preference 2 no-reg then i shut down the serial interface and the phones register to srts router, then I no shut the ser interface and the phones register back to cucm. At this point i do: sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000 voip up up 3...$0syst ipv4:140.50.64.20 3001 voip up up 3...$1syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up down2 50/0/1 20002 pots up down2 50/0/2 The ephone-dns dial-peer created are down and with no des pattern. After some changes to telephony-ser configuration when the phones go to srst and then back to cucm I have the following result: #sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000voip up up 3...$ 0 syst ipv4:140.50.64.20 3001voip up up 3...$ 1 syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up up 3002$ 2 50/0/1 20002 pots up up 3001$ 2 50/0/2 So when i try to call ephone 3001 let say from pstn the call fails becouse dial-peer 20002 is a closer match than 3...$ (dial peer 3001 poiting to CUCM), I know that I can chage this behaviour with dial-peer hunt ? 0-7 Dial-peer hunting choices, listed in hunting order within each choice: 0 - Longest match in phone number, explicit preference, random selection. 1 - Longest match in phone number, explicit preference, least recent use. 2 - Explicit preference, longest match in phone number, random selection. 3 - Explicit preference, longest match in phone number, least recent use. 4 - Least recent use, longest match in phone number, explicit preference. 5 - Least recent use, explicit preference, longest match in phone number. 6 - Random selection. 7 - Least recent use. dial-peer hunt 3 This way explicit preference will be checked before, but my question is: What changes in cme as srst make dial-peer to be persistent when the gw is NOT in srst? I think that this behaviour is the result of create cnf command but I'm not sure Thanks in advance Hotmail: Free, trusted and rich email service. Get it now. Hotmail: Trusted email with powerful SPAM protection. Sign up now. _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Help with + Dialing Question
Ken, That is the expected behavior when transferring incoming pstn calls to other internal (configured with localization/cg xform patterns css) phones, Please take a look at the following: http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html#wp1266646 hth, On Thu, Apr 8, 2010 at 6:35 PM, Beck, Ken kb...@vectorusa.com wrote: Direct inbound dialing from the PSTN displays correctly on the phone and in missed/received calls, however if I dial the AA built in CUC and dial an extension; the phone displays +1 and the number instead of just the ten digits. Can someone help me out where to look? Essentially the call is coming from the VM-ports and everything seems to be configured correctly. Thanks, Ken Information contained in this e-mail message is intended only for the individual to whom it is addressed and is private and confidential. If you are not the intended recipient, or the employee or agent responsible for delivering this message to the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you are not the intended recipient of this e-mail, please kindly destroy it and notify the sender immediately by reply e-mail. Thank you for your cooperation. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CCX NTP
Hi: I've configured NTP (hq router) at CCX Cisco Unified CM Configuration page, but doing some tests with time of day script option show me that the script was taken the time from windows clock... Is this correct? How can I verify that CCX takes the time from the NTP and that it is synch wit it? Thanks _ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME in SRST dial-peer issue
The other problem that you describe will be fixed by a reboot of the CME GW. I've seen that to, but in my case I had added a cor list to the ephone. It gave me exactly the same sort of problem that you describe. There is another fix for this and that's to remove all config for affected the ephone-dn,. Copy paste before to notepad, then add it back again. Don't forget to reapply the dn to the ephone with the button command. This worked for me when I ran into this quirky behavior, that is before I knew that a simple reboot would also do it. :) Roger Källberg Unified Communication Consultant Cygate AB From: Angel Perez [mailto:gorr...@hotmail.com] Sent: den 9 april 2010 11:48 To: Roger Källberg; osl osl Cc: vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi: Yes it could be a bug, becouse from my experience the behaviour is not uniform with this issue Also when I change ephone name or label somentimes the ephone doesn't get a dn and it register without dn... another strange issue too I'm running IOS 12.4(20)T Thanks From: roger.kallb...@cygate.se To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com CC: vma...@ipexpert.com Date: Fri, 9 Apr 2010 10:32:12 +0200 Subject: RE: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi, If I'm not completely wrong I do believe that Vik mentioned this at the ILT in March. As I can recall it's a bug in the CME version that we have in the lab or possibly it's IOS related. Maybe Vik can verify/clarify this? Roger Källberg Unified Communication Consultant Cygate AB From: Angel Perez [mailto:gorr...@hotmail.com] Sent: den 9 april 2010 09:53 To: osl osl Subject: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi all: I've the following issue with cme as srst: When I've configured telephony-service srst mode auto-provision all srst dn line-mode octo max-ephones 2 max-dn 20 preference 2 no-reg then i shut down the serial interface and the phones register to srts router, then I no shut the ser interface and the phones register back to cucm. At this point i do: sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000 voip up up 3...$0syst ipv4:140.50.64.20 3001 voip up up 3...$1syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up down2 50/0/1 20002 pots up down2 50/0/2 The ephone-dns dial-peer created are down and with no des pattern. After some changes to telephony-ser configuration when the phones go to srst and then back to cucm I have the following result: #sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000voip up up 3...$ 0 syst ipv4:140.50.64.20 3001voip up up 3...$ 1 syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up up 3002$ 2 50/0/1 20002 pots up up 3001$ 2 50/0/2 So when i try to call ephone 3001 let say from pstn the call fails becouse dial-peer 20002 is a closer match than 3...$ (dial peer 3001 poiting to CUCM), I know that I can chage this behaviour with dial-peer hunt ? 0-7 Dial-peer hunting choices, listed in hunting order within each choice: 0 - Longest match in phone number, explicit preference, random selection. 1 - Longest match in phone number, explicit preference, least recent use. 2 - Explicit preference, longest match in phone number, random selection. 3 - Explicit preference, longest match in phone number, least recent use. 4 - Least recent use, longest match in phone number, explicit preference. 5 - Least recent use, explicit preference, longest match in phone number. 6 - Random selection. 7 - Least recent use. dial-peer hunt 3 This way explicit preference will be checked before, but my question is: What changes in cme as srst make dial-peer to be persistent when the gw is NOT in srst? I think that this behaviour is the result of create cnf command but I'm not sure Thanks in advance Hotmail: Free, trusted and rich email service. Get it now.https://signup.live.com/signup.aspx?id=60969 Hotmail: Trusted email with powerful SPAM protection. Sign up
[OSL | CCIE_Voice] Can't remove match access-group from class-map after auto qos
I'm having some problems modifying the class-map and policy-map after running autoqos. I am revising Vol 1 lab 10, and running auto qos voip fr-atm under the subinterface DLCI towards BR2: interface Serial0/0/0:0.200 point-to-point bandwidth 768 snmp trap link-status frame-relay interface-dlci 200 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/0:0-200 auto qos voip fr-atm This provisions the standard access-lists, class-maps and policy-maps as well as the virtual-template. I can modify the policy-map, but for some reason the command no class AutoQoS-VoIP-Remark will not remove that section from the policy-map and also I cannot remove the ACLs from class-map match-any AutoQoS-VoIP-Control-UnTrust and class-map match-any AutoQoS-VoIP-RTP-UnTrust If I remove the ACLs first, then try to remove them from the class-map, the router crashes and reloads. Is there some sort of IOS/auto qos bug I'm hitting? I'm running: Cisco IOS Software, 2800 Software (C2800NM-ADVENTERPRISEK9_IVS_LI-M), Version 12.4(22)T4, RELEASE SOFTWARE (fc2) class-map match-any AutoQoS-VoIP-Remark match ip dscp ef match ip dscp cs3 match ip dscp af31 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 priority 116 set dscp ef class AutoQoS-VoIP-Control-UnTrust bandwidth 65 set dscp cs3 class AutoQoS-VoIP-Remark class class-default fair-queue police rate percent 65 exceed-action set-dscp-transmit default Thank you in advance! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM GK trunk keeps unregistering?
This seems to have been caused by performance issues running all my UC servers in VMware server 2/Ubuntu 64. Today with only the pub running, the CUCM trunk stays up. After my first lab attempt (soon) I'm going to rebuild using ESXi4 and see how that goes... On Thu, Apr 8, 2010 at 2:08 PM, Stephen Greszczyszyn sgres...@gmail.com wrote: I'm doing a simple GK setup where the CUCM is registering a GK trunk to HQ and BR2 as well. After both BR2 and HQ trunks are registered, I can make calls between sites. Then after some testing I get a busy signal, and when I check I see that the CUCM GK trunk has unregistered and I need to reset the Trunk in CUCM to force it to re-register. Any ideas? Thanks in advance... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 - MWI
Omotayo, When integrating CUCME and CUE it is best to use sip-notify vs. rtp-nte for dtmf configured on the dial-peer. dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay sip-notify codec g711ulaw no vad The dtmf-relay rtp-nte command sets the SIP DTMF relay mechanism to use the media path to relay incoming and outgoing DTMF signals to Cisco Unity Express. The dtmf-relay sip-notify command sets the SIP DTMF relay mechanism to use Unsolicited-Notify messages to relay incoming and outgoing DTMF signals. HTH, Amy --- Amy Ryan CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 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, 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: Omotayo adefilabi...@gmail.com Date: Fri, 9 Apr 2010 09:40:36 +0100 To: Roger Källberg roger.kallb...@cygate.se Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab 1 - MWI dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay rtp-nte codec g711ulaw no vad ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME in SRST dial-peer issue
Yes, a reboot worked for me too another workaround is: telephony-service srst mode auto-provision none activate srst fallback to cucm telephony-service srst mode auto-provision all activate srst But I aggree with you that reboot could be faster (and you can continue with other question) thanks From: roger.kallb...@cygate.se To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com CC: vma...@ipexpert.com Date: Fri, 9 Apr 2010 13:49:16 +0200 Subject: RE: [OSL | CCIE_Voice] CME in SRST dial-peer issue The other problem that you describe will be “fixed” by a reboot of the CME GW. I’ve seen that to, but in my case I had added a cor list to the ephone. It gave me exactly the same sort of problem that you describe. There is another “fix” for this and that’s to remove all config for affected the ephone-dn,. Copy paste before to notepad, then add it back again. Don’t forget to reapply the dn to the ephone with the button command. This worked for me when I ran into this quirky behavior, that is before I knew that a simple reboot would also do it. J Roger Källberg Unified Communication Consultant Cygate AB From: Angel Perez [mailto:gorr...@hotmail.com] Sent: den 9 april 2010 11:48 To: Roger Källberg; osl osl Cc: vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi: Yes it could be a bug, becouse from my experience the behaviour is not uniform with this issue Also when I change ephone name or label somentimes the ephone doesn't get a dn and it register without dn... another strange issue too I'm running IOS 12.4(20)T Thanks From: roger.kallb...@cygate.se To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com CC: vma...@ipexpert.com Date: Fri, 9 Apr 2010 10:32:12 +0200 Subject: RE: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi, If I’m not completely wrong I do believe that Vik mentioned this at the ILT in March. As I can recall it’s a bug in the CME version that we have in the lab or possibly it’s IOS related. Maybe Vik can verify/clarify this? Roger Källberg Unified Communication Consultant Cygate AB From: Angel Perez [mailto:gorr...@hotmail.com] Sent: den 9 april 2010 09:53 To: osl osl Subject: [OSL | CCIE_Voice] CME in SRST dial-peer issue Hi all: I've the following issue with cme as srst: When I've configured telephony-service srst mode auto-provision all srst dn line-mode octo max-ephones 2 max-dn 20 preference 2 no-reg then i shut down the serial interface and the phones register to srts router, then I no shut the ser interface and the phones register back to cucm. At this point i do: sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000 voip up up 3...$0syst ipv4:140.50.64.20 3001 voip up up 3...$1syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up down2 50/0/1 20002 pots up down2 50/0/2 The ephone-dns dial-peer created are down and with no des pattern. After some changes to telephony-ser configuration when the phones go to srst and then back to cucm I have the following result: #sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 3000voip up up 3...$ 0 syst ipv4:140.50.64.20 3001voip up up 3...$ 1 syst ipv4:140.50.64.21 999 pots up up 999 0 up 0/0/0:15 20001 pots up up 3002$ 2 50/0/1 20002 pots up up 3001$ 2 50/0/2 So when i try to call ephone 3001 let say from pstn the call fails becouse dial-peer 20002 is a closer match than 3...$ (dial peer 3001 poiting to CUCM), I know that I can chage this behaviour with dial-peer hunt ? 0-7 Dial-peer hunting choices, listed in hunting order within each choice: 0 - Longest match in phone number, explicit preference, random selection. 1 - Longest match in phone number, explicit preference, least recent use. 2 - Explicit preference, longest match in phone number, random selection. 3 - Explicit preference, longest match in phone number, least recent use. 4 - Least recent use, longest match in phone number, explicit preference. 5 - Least recent use, explicit preference, longest match in phone number. 6 - Random
Re: [OSL | CCIE_Voice] Can't remove match access-group from class-map after auto qos
Scratch that, I realised that the serial interface was not up to BR2 and I remember that being one of the requirements for auto qos? Anyway, I am now able to modify the class-maps and policy-maps. On Fri, Apr 9, 2010 at 1:08 PM, Stephen Greszczyszyn sgres...@gmail.com wrote: I'm having some problems modifying the class-map and policy-map after running autoqos. I am revising Vol 1 lab 10, and running auto qos voip fr-atm under the subinterface DLCI towards BR2: ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM GK trunk keeps unregistering?
Hi: I've seen problems with gk trunk registrations/unregistration before, in my case a problem with ntp server was behind the issue, an incorrect time stamp can cause the trunk to unregister, is your gk router and ccm synch with ntp server? hth Date: Fri, 9 Apr 2010 13:11:28 +0100 From: sgres...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CUCM GK trunk keeps unregistering? This seems to have been caused by performance issues running all my UC servers in VMware server 2/Ubuntu 64. Today with only the pub running, the CUCM trunk stays up. After my first lab attempt (soon) I'm going to rebuild using ESXi4 and see how that goes... On Thu, Apr 8, 2010 at 2:08 PM, Stephen Greszczyszyn sgres...@gmail.com wrote: I'm doing a simple GK setup where the CUCM is registering a GK trunk to HQ and BR2 as well. After both BR2 and HQ trunks are registered, I can make calls between sites. Then after some testing I get a busy signal, and when I check I see that the CUCM GK trunk has unregistered and I need to reset the Trunk in CUCM to force it to re-register. Any ideas? Thanks in advance... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Help with + Dialing Question
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html#wp1266646 For those without a partner account, here is the link and corresponding text: Globalizing and Localizing Calling Party Numbers for Transferred Calls /*The transfer feature relies on mid-call updates, so depending on the scenario, a transferred call may not support globalization and localization of the calling party number.*/ (Calling party normalization supports globalization and localization during call setup for each hop of the call, not for mid-call updates.) For examples of how calling party normalization works for transferred calls, see the following sections: .Calling Party Normalization for On Net Transferred Call Across a Gateway .Calling Party Normalization for Transferred Call Through an Incoming Gateway Calling Party Normalization for On Net Transferred Call Across a Gateway Phone A with extension 12345 and phone number of 972 500 2345 calls Phone B with extension 54321 and phone number 972 500 4321; when the call arrives on extension 54321, calling party number 12345 displays on Phone B. Phone B transfers the call to Phone C in San Jose through a San Jose gateway. During the initiation of the transfer, Phone C displays the calling party number for Phone B as 972 500 4321. After the transfer completes, Phone C displays the calling party number for Phone A as 12345. Matthew Berry /A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written/ *Gmail:* ciscovoiceguru *Skype:* ciscovoiceguru *Twitter:* ciscovoiceguru *1st Lab Attempt: *Aug 16, 2010 On 4/9/2010 6:33 AM, Otto Sanchez wrote: Ken, That is the expected behavior when transferring incoming pstn calls to other internal (configured with localization/cg xform patterns css) phones, Please take a look at the following: http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html#wp1266646 hth, On Thu, Apr 8, 2010 at 6:35 PM, Beck, Ken kb...@vectorusa.com mailto:kb...@vectorusa.com wrote: Direct inbound dialing from the PSTN displays correctly on the phone and in missed/received calls, however if I dial the AA built in CUC and dial an extension; the phone displays +1 and the number instead of just the ten digits. Can someone help me out where to look? Essentially the call is coming from the VM-ports and everything seems to be configured correctly. Thanks, Ken Information contained in this e-mail message is intended only for the individual to whom it is addressed and is private and confidential. If you are not the intended recipient, or the employee or agent responsible for delivering this message to the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you are not the intended recipient of this e-mail, please kindly destroy it and notify the sender immediately by reply e-mail. Thank you for your cooperation. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: 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
[OSL | CCIE_Voice] Unity Connection Timezones
Can someone tell me how the timezones relate for Unity Connection 1) Timezone configured in CLI (change requires reboot) versus 2) UC Admin System Settings General Config Time Zone (change requires no reboot) Thanks, Ken ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCX NTP
Try this ... http://docs.google.com/viewer?a=vq=cache:OwhhDwCTWvUJ:www.ciscointernethome.net/application/pdf/paws/21003/NTP.pdf+setting+ntp+on+cisco+unityhl=engl=uspid=blsrcid=ADGEEShBypdSUSMrKB7TKj1CYAmkFmcjyAtpdPjeUGZ8uK-Ra5U5isoPjUfbD0WCUISmYqyAk520lxEcl1n3aQDe9MsMUzb6yQGyLsO7K8N-vhGCnnxi86Lw7hrjfK--RQvQpQI5PwnYsig=AHIEtbQlYWpEQBlE6qGVycOxLi3Y-9hUFw On Fri, Apr 9, 2010 at 4:48 AM, Angel Perez gorr...@hotmail.com wrote: Hi: I've configured NTP (hq router) at CCX Cisco Unified CM Configuration page, but doing some tests with time of day script option show me that the script was taken the time from windows clock... Is this correct? How can I verify that CCX takes the time from the NTP and that it is synch wit it? Thanks -- Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®.http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009 ___ 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] Lab 1 - MWI
Omotayo, Amy you are right about dtmf-relay sip-notify but I think he also has something ls which doesn't match. The best practice is that always you need to check your configs at both ends in this case CME and CUE . So could you post the relevant MWI configs from CUE too ? Without knowing your relevant MWI config from CUE , I think that the DN (incoming called-number) for MWI in CUE doesn't match the CME MWI. Take a look at my sample bellow from a real world project about 2 weeks ago on CME/CUE: CME: dial-peer voice 1001 voip description MWI Inbound Dial-peer destination-pattern ^100[12]$ session protocol sipv2 session target ipv4:172.16.64.20 incoming called-number 100[12] dtmf-relay sip-notify codec g711ulaw no vad CUE: ccn trigger sip phonenumber 1000 application voicemail enabled maxsessions 8 end trigger .. voicemail broadcast mwi voicemail callerid voicemail default mailboxsize 10600 voicemail broadcast recording time 300 voicemail default messagesize 280 Summary , in this case incoming called-number 100[12] match CUE VM Extn 1000. hth ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 - MWI
thanks, i will check all this and have a feedback On Fri, Apr 9, 2010 at 5:36 PM, Mike Peterson polobi...@yahoo.com wrote: Omotayo, Amy you are right about dtmf-relay sip-notify but I think he also has something ls which doesn't match. The best practice is that always you need to check your configs at both ends in this case CME and CUE . So could you post the relevant MWI configs from CUE too ? Without knowing your relevant MWI config from CUE , I think that the DN (incoming called-number) for MWI in CUE doesn't match the CME MWI. Take a look at my sample bellow from a real world project about 2 weeks ago on CME/CUE: CME: dial-peer voice 1001 voip description MWI Inbound Dial-peer destination-pattern ^100[12]$ session protocol sipv2 session target ipv4:172.16.64.20 incoming called-number 100[12] dtmf-relay sip-notify codec g711ulaw no vad CUE: ccn trigger sip phonenumber 1000 application voicemail enabled maxsessions 8 end trigger .. voicemail broadcast mwi voicemail callerid voicemail default mailboxsize 10600 voicemail broadcast recording time 300 voicemail default messagesize 280 Summary , in this case incoming called-number 100[12] match CUE VM Extn 1000. hth ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 - MWI
Are you usig voice class codec in default Incoming voip dial-peer? Put only g729r8 in incoming voip dial-peer and remove voice class codec. Ash On 09/04/2010 13:18, Amy Ryan wrote: Omotayo, When integrating CUCME and CUE it is best to use sip-notify vs. rtp-nte for dtmf configured on the dial-peer. dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] * dtmf-relay sip-notify * codec g711ulaw no vad The dtmf-relay rtp-nte command sets the SIP DTMF relay mechanism to use the media path to relay incoming and outgoing DTMF signals to Cisco Unity Express. The dtmf-relay sip-notify command sets the SIP DTMF relay mechanism to use Unsolicited-Notify messages to relay incoming and outgoing DTMF signals. HTH, Amy --- Amy Ryan -- CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: _ar...@ipexpert.com _Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat _http://www.ipexpert.com/chat_ eFax: +1.810.454.0130 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, 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: *Omotayo adefilabi...@gmail.com *Date: *Fri, 9 Apr 2010 09:40:36 +0100 *To: *Roger Källberg roger.kallb...@cygate.se *Cc: *OSL Group ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] Lab 1 - MWI dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay rtp-nte codec g711ulaw no vad ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Thanks, Ashar Siddiqui ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Codec command in voice register pool
Is it MUST to give codec g711ulaw voice register pool , if we don't give what to do we break pls ??? thanks for your help... -M ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
