Re: [OSL | CCIE_Voice] SRST transfer system and pattern
Thanks, makes sense. One of those few configurations on the exam that sticks to the design guidelines field deployments. :) :) Date: Sat, 16 Feb 2013 17:48:16 -0600 Subject: Re: [OSL | CCIE_Voice] SRST transfer system and pattern From: ramcharan.a...@gmail.com To: corygray22...@hotmail.com CC: pixarperf...@live.com; ccie_voice@onlinestudylist.com Hi, As per cisco CME design guide these commands are necessary. Please refer cisco CME SRND. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/clproc.html#wp1068396 Regards, Ramcharan Arya CCIE # 28926 ( RS) On Fri, Feb 15, 2013 at 4:51 PM, Cory Gray corygray22...@hotmail.com wrote: I have had several conversations with people on this. Everyone can easily make SRST work but scoring points seems to be the trickiest thing in the lab. So I do not think anyone knows for sure what should or should not be on the “template” I have never scored any points there so I cannot give an OPINION on what should or should not be there. People say they score points and then go with the same template on the next lab and get 0 so it is a mystery. People can share templates without breaking NDA since the question is never discussed. Getting the question right is the easy part! From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pixar Perfect Sent: Friday, February 15, 2013 5:26 PM To: CCIE Voice OSL Subject: [OSL | CCIE_Voice] SRST transfer system and pattern transfer-system full-consultdo we need to specify this? I thought by default it is wnabled but I read on voiceie forum someone scored nothing on SRST adn the only conclusion was the transfersystem consult was missing. Any thoughts? srst mode auto-provision all srst ephone description SRST-EPHONES-CME srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 preference 2 no-reg both ip source-address 10.10.1.13 SiteC Loopback port 2000 time-zone 42 max-conferences 8 gain -6 call-forward pattern .T time-webedit transfer-system full-consult transfer-pattern .T ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] cBarge Barge softkey
When working on Shared DNs and cBarge question (5-lab handbook, Lab1) that needs use of CFB on Site C, do we need to remove the Barge Softkey from the Remote in Use state? do you think it is good idea to disable Built in Bridge for the two phones that have a shared line and need GW CFB for conferencing.? the solution guide has an example that has the Barge softkey left there in Remote In Use. Per IPEXPERT's bootcamp, the recommendation was not to tamper with the existing Softkey layout and keep adding softkeys. It makes sense however, this particular Barge vs cBarge is tricky thing ... i would be least worried abt these things but it will be unfortunate if the script is looking for Barge softkey as well :) ... the notorious grading script process worries me as it is the deal breaker :) thx...pixar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cBarge Barge softkey
You have a right to worry. Default is default setting for built-in bridge. The default is off. Barge is part of default remote in use. I would add cbarge and not mess with anything else UNLESS explicitly told or some wording points you in that direction. Sent from my iPhone On Feb 17, 2013, at 6:01 AM, Pixar Perfect pixarperf...@live.com wrote: When working on Shared DNs and cBarge question (5-lab handbook, Lab1) that needs use of CFB on Site C, do we need to remove the Barge Softkey from the Remote in Use state? do you think it is good idea to disable Built in Bridge for the two phones that have a shared line and need GW CFB for conferencing.? the solution guide has an example that has the Barge softkey left there in Remote In Use. Per IPEXPERT's bootcamp, the recommendation was not to tamper with the existing Softkey layout and keep adding softkeys. It makes sense however, this particular Barge vs cBarge is tricky thing ... i would be least worried abt these things but it will be unfortunate if the script is looking for Barge softkey as well :) ... the notorious grading script process worries me as it is the deal breaker :) thx...pixar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE Dropped Calls
I have attached the config for you to reference. I would provide some debugs, but it isn't failing at all today. No config changes except to cp tones this morning. Just want to make sure i'm not blatantly missing something... Config r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910 translate calling 9101 translate called 910 ! voice translation-profile 911 translate calling 9111 translate called 911 ! voice translation-profile 97 translate calling 971 translate called 97 ! voice translation-profile strip translate called 1 ! ! voice-card 0 dsp services dspfarm ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 ! controller E1 0/1/1 ! controller T1 1/0 cablelength long 0db ! controller T1 1/1 cablelength long 0db ! ! ! ! ! interface Loopback0 ip address 192.168.96.2 255.255.255.255 h323-gateway voip bind srcaddr 192.168.96.2 ! interface GigabitEthernet0/0 no ip address duplex auto speed auto ! interface GigabitEthernet0/0.105 encapsulation dot1Q 105 native ip address 192.168.105.1 255.255.255.0 ! interface GigabitEthernet0/0.106 encapsulation dot1Q 106 ip address 192.168.106.1 255.255.255.0 ! interface Service-Engine0/0 ip unnumbered GigabitEthernet0/0.106 service-module ip address 192.168.106.2 255.255.255.0 service-module ip default-gateway 192.168.106.1 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface FastEthernet0/3/0 shutdown ! interface FastEthernet0/3/1 shutdown ! interface FastEthernet0/3/2 shutdown ! interface FastEthernet0/3/3 shutdown ! interface Serial0/0/0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/0.1 point-to-point description FR-WAN INTERFACE - DLCI 102 ip address 192.168.111.10 255.255.255.252 shutdown frame-relay interface-dlci 102 ip rsvp bandwidth 64 ! interface Serial0/0/1 no ip address shutdown clock rate 200 ! interface Serial0/1/0:15 no ip address encapsulation hdlc isdn switch-type
[OSL | CCIE_Voice] Custom Tones
All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910 translate calling 9101 translate called 910 ! voice translation-profile 911 translate calling 9111 translate called 911 ! voice translation-profile 97 translate calling 971 translate called 97 ! voice translation-profile strip translate called 1 ! ! voice-card 0 dsp services dspfarm ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 ! controller E1 0/1/1 ! controller T1 1/0 cablelength long 0db ! controller T1 1/1 cablelength long 0db ! ! ! ! ! interface Loopback0 ip address 192.168.96.2 255.255.255.255 h323-gateway voip bind srcaddr 192.168.96.2 ! interface GigabitEthernet0/0 no ip address duplex auto speed auto ! interface GigabitEthernet0/0.105 encapsulation dot1Q 105 native ip address 192.168.105.1 255.255.255.0 ! interface GigabitEthernet0/0.106 encapsulation dot1Q 106 ip address 192.168.106.1 255.255.255.0 ! interface Service-Engine0/0 ip unnumbered GigabitEthernet0/0.106 service-module ip address 192.168.106.2 255.255.255.0 service-module ip default-gateway 192.168.106.1 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface FastEthernet0/3/0 shutdown ! interface FastEthernet0/3/1 shutdown ! interface FastEthernet0/3/2 shutdown ! interface FastEthernet0/3/3 shutdown ! interface Serial0/0/0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/0.1 point-to-point description FR-WAN INTERFACE - DLCI 102 ip address 192.168.111.10 255.255.255.252 shutdown frame-relay interface-dlci 102 ip rsvp bandwidth 64 !
Re: [OSL | CCIE_Voice] SRST transfer system and pattern
I'm adding secondary dialtone to my CUCME and SRST configurations as well. In my mind, we should be trying to preserve as much of the CUCM configuration as possible. Not sure that it helps with grading, but better safe than sorry I guess. On Sun, Feb 17, 2013 at 4:58 AM, Pixar Perfect pixarperf...@live.comwrote: Thanks, makes sense. One of those few configurations on the exam that sticks to the design guidelines field deployments. :) :) -- Date: Sat, 16 Feb 2013 17:48:16 -0600 Subject: Re: [OSL | CCIE_Voice] SRST transfer system and pattern From: ramcharan.a...@gmail.com To: corygray22...@hotmail.com CC: pixarperf...@live.com; ccie_voice@onlinestudylist.com Hi, As per cisco CME design guide these commands are necessary. Please refer cisco CME SRND. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/clproc.html#wp1068396 Regards, Ramcharan Arya CCIE # 28926 ( RS) On Fri, Feb 15, 2013 at 4:51 PM, Cory Gray corygray22...@hotmail.comwrote: I have had several conversations with people on this. Everyone can easily make SRST work but scoring points seems to be the trickiest thing in the lab. So I do not think anyone knows for sure what should or should not be on the “template” I have never scored any points there so I cannot give an OPINION on what should or should not be there. People say they score points and then go with the same template on the next lab and get 0 so it is a mystery. People can share templates without breaking NDA since the question is never discussed. Getting the question right is the easy part! ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Pixar Perfect *Sent:* Friday, February 15, 2013 5:26 PM *To:* CCIE Voice OSL *Subject:* [OSL | CCIE_Voice] SRST transfer system and pattern ** ** transfer-system full-consultdo we need to specify this? I thought by default it is wnabled but I read on voiceie forum someone scored nothing on SRST adn the only conclusion was the transfersystem consult was missing. Any thoughts? ** ** srst mode auto-provision all srst ephone description SRST-EPHONES-CME srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 preference 2 no-reg both ip source-address 10.10.1.13 SiteC Loopback port 2000 time-zone 42 max-conferences 8 gain -6 call-forward pattern .T time-webedit * transfer-system full-consult* * transfer-pattern .T* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Custom Tones
I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910 translate calling 9101 translate called 910 ! voice translation-profile 911 translate calling 9111 translate called 911 ! voice translation-profile 97 translate calling 971 translate called 97 ! voice translation-profile strip translate called 1 ! ! voice-card 0 dsp services dspfarm ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 ! controller E1 0/1/1 ! controller T1 1/0 cablelength long 0db ! controller T1 1/1 cablelength long 0db ! ! ! ! ! interface Loopback0 ip address 192.168.96.2 255.255.255.255 h323-gateway voip bind srcaddr 192.168.96.2 ! interface GigabitEthernet0/0 no ip address duplex auto speed auto ! interface GigabitEthernet0/0.105 encapsulation dot1Q 105 native ip address 192.168.105.1 255.255.255.0 ! interface GigabitEthernet0/0.106 encapsulation dot1Q 106 ip address 192.168.106.1 255.255.255.0 ! interface Service-Engine0/0 ip unnumbered GigabitEthernet0/0.106 service-module ip address 192.168.106.2 255.255.255.0
Re: [OSL | CCIE_Voice] Custom Tones
I haven't tested this recently, but it may help to make the join/leave tones use different frequencies, as well as using different time intervals for the cadence. I'm not sure why you're getting these strange results (two tones on join when your cadence only shows one and no tone on leave), but there may be some strange feature (or bug) that has to do with both join and leave using the same frequency. voice class custom-cptone leave dualtone conference frequency 300 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 cadence 800 -Justin On Sun, Feb 17, 2013 at 1:56 PM, William Bell b...@ucguerrilla.com wrote: I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910 translate calling 9101 translate called 910 ! voice translation-profile 911 translate calling 9111 translate called 911 ! voice translation-profile 97 translate calling 971 translate called 97 ! voice translation-profile strip translate called 1 ! ! voice-card 0 dsp services dspfarm ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 ! controller E1
[OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 177.1.10.10 1720 177.1.10.10 32811 CUCM VOIP-GW H323-ID: CUCM_TRUNK_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 32788 CUCM VOIP-GW H323-ID: CUCM_TRUNK_2 Voice Capacity Max.= Avail.= Current.= 0 177.1.254.3 1720 177.1.254.3 63360 CUCM H323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CorpHQ(config-dial-peer)# CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 31* Zone CUCM master gateway list: 177.1.254.3:1720 CUCME Prefix: 1* Zone CUCM master gateway list: 177.1.10.10:1720 CUCM_TRUNK_1 177.1.10.20:1720 CUCM_TRUNK_2 CorpHQ(config-dial-peer)# ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 177.1.10.10 1720 177.1.10.10 32811 CUCM VOIP-GW H323-ID: CUCM_TRUNK_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 32788 CUCM VOIP-GW H323-ID: CUCM_TRUNK_2 Voice Capacity Max.= Avail.= Current.= 0 177.1.254.3 1720 177.1.254.3 63360 CUCM H323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CorpHQ(config-dial-peer)# CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 31* Zone CUCM master gateway list: 177.1.254.3:1720 CUCME Prefix: 1* Zone CUCM master gateway list: 177.1.10.10:1720 CUCM_TRUNK_1 177.1.10.20:1720 CUCM_TRUNK_2 CorpHQ(config-dial-peer)# ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
I did that and allow connections as well On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com wrote: Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 177.1.10.10 1720 177.1.10.10 32811 CUCM VOIP-GW H323-ID: CUCM_TRUNK_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 32788 CUCM VOIP-GW H323-ID: CUCM_TRUNK_2 Voice Capacity Max.= Avail.= Current.= 0 177.1.254.3 1720 177.1.254.3 63360 CUCM H323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CorpHQ(config-dial-peer)# CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 31* Zone CUCM master gateway list: 177.1.254.3:1720 CUCME Prefix: 1* Zone CUCM master gateway list: 177.1.10.10:1720 CUCM_TRUNK_1 177.1.10.20:1720 CUCM_TRUNK_2 CorpHQ(config-dial-peer)# ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
Should not have allow connections either unless you are doing cube but that should not break it. Debug h22r ans1 and look to see if there is detail on why the call is failing. Make sure you are using g729 as well Sent from my iPhone On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I did that and allow connections as well On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com wrote: Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 177.1.10.10 1720 177.1.10.10 32811 CUCM VOIP-GW H323-ID: CUCM_TRUNK_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 32788 CUCM VOIP-GW H323-ID: CUCM_TRUNK_2 Voice Capacity Max.= Avail.= Current.= 0 177.1.254.3 1720 177.1.254.3 63360 CUCM H323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CorpHQ(config-dial-peer)# CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 31* Zone CUCM master gateway list: 177.1.254.3:1720 CUCME Prefix: 1* Zone CUCM master gateway list: 177.1.10.10:1720 CUCM_TRUNK_1 177.1.10.20:1720 CUCM_TRUNK_2 CorpHQ(config-dial-peer)# ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
Yes i am using g729 and i configured them from both sides CUCM side as region and location /devicepool and voice class codec as cme side. I am able to send calls from CME to CUCM but cucm unable to place calls to CME On Feb 17, 2013, at 3:51 PM, Cory Gray corygray22...@hotmail.com wrote: Should not have allow connections either unless you are doing cube but that should not break it. Debug h22r ans1 and look to see if there is detail on why the call is failing. Make sure you are using g729 as well Sent from my iPhone On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I did that and allow connections as well On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com wrote: Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 177.1.10.10 1720 177.1.10.10 32811 CUCM VOIP-GW H323-ID: CUCM_TRUNK_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 32788 CUCM VOIP-GW H323-ID: CUCM_TRUNK_2 Voice Capacity Max.= Avail.= Current.= 0 177.1.254.3 1720 177.1.254.3 63360 CUCM H323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CorpHQ(config-dial-peer)# CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 31* Zone CUCM master gateway list: 177.1.254.3:1720 CUCME Prefix: 1* Zone CUCM master gateway list: 177.1.10.10:1720 CUCM_TRUNK_1 177.1.10.20:1720 CUCM_TRUNK_2 CorpHQ(config-dial-peer)# ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
I am sorry. I had it backwards. I thought you had an issue routing to CUCM. For call into CUCME, you need this Dial peer voice 3000 voip Incoming called ^3...$ Dtmf-r h245a No vad Translation-profile in STRIP ! Voice translation-rule 1 Rule 1 /.+\(\)$/ /\1/ ! Voice translation-profile STRIP Translate called 1 -Original Message- From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com] Sent: Sunday, February 17, 2013 5:56 PM To: Cory Gray Cc: ccie_voice Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue Yes i am using g729 and i configured them from both sides CUCM side as region and location /devicepool and voice class codec as cme side. I am able to send calls from CME to CUCM but cucm unable to place calls to CME On Feb 17, 2013, at 3:51 PM, Cory Gray corygray22...@hotmail.com wrote: Should not have allow connections either unless you are doing cube but that should not break it. Debug h22r ans1 and look to see if there is detail on why the call is failing. Make sure you are using g729 as well Sent from my iPhone On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I did that and allow connections as well On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com wrote: Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 177.1.10.10 1720 177.1.10.10 32811 CUCM VOIP-GW H323-ID: CUCM_TRUNK_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 32788 CUCM VOIP-GW H323-ID: CUCM_TRUNK_2 Voice Capacity Max.= Avail.= Current.= 0 177.1.254.3 1720 177.1.254.3 63360 CUCM H323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CorpHQ(config-dial-peer)# CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 31* Zone CUCM master gateway list: 177.1.254.3:1720 CUCME Prefix: 1* Zone CUCM master gateway list: 177.1.10.10:1720 CUCM_TRUNK_1 177.1.10.20:1720 CUCM_TRUNK_2 CorpHQ(config-dial-peer)# ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
Thank you so much for your efforts. I believe it may need a strip but i don't know exactly what or how to strip the prefix as with CUBE it works without need for translation rule. Thanks for info i will try and feed you back. Thanks, Hesham On Feb 17, 2013, at 4:08 PM, Cory Gray corygray22...@hotmail.com wrote: I am sorry. I had it backwards. I thought you had an issue routing to CUCM. For call into CUCME, you need this Dial peer voice 3000 voip Incoming called ^3...$ Dtmf-r h245a No vad Translation-profile in STRIP ! Voice translation-rule 1 Rule 1 /.+\(\)$/ /\1/ ! Voice translation-profile STRIP Translate called 1 -Original Message- From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com] Sent: Sunday, February 17, 2013 5:56 PM To: Cory Gray Cc: ccie_voice Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue Yes i am using g729 and i configured them from both sides CUCM side as region and location /devicepool and voice class codec as cme side. I am able to send calls from CME to CUCM but cucm unable to place calls to CME On Feb 17, 2013, at 3:51 PM, Cory Gray corygray22...@hotmail.com wrote: Should not have allow connections either unless you are doing cube but that should not break it. Debug h22r ans1 and look to see if there is detail on why the call is failing. Make sure you are using g729 as well Sent from my iPhone On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I did that and allow connections as well On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com wrote: Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 177.1.10.10 1720 177.1.10.10 32811 CUCM VOIP-GW H323-ID: CUCM_TRUNK_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 32788 CUCM VOIP-GW H323-ID: CUCM_TRUNK_2 Voice Capacity Max.= Avail.= Current.= 0 177.1.254.3 1720 177.1.254.3 63360 CUCM H323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CorpHQ(config-dial-peer)# CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 31* Zone CUCM master gateway list: 177.1.254.3:1720 CUCME Prefix: 1* Zone CUCM master gateway list: 177.1.10.10:1720 CUCM_TRUNK_1 177.1.10.20:1720 CUCM_TRUNK_2 CorpHQ(config-dial-peer)# ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
With CUBE, there is no tech prefix so that is why you don't need it here. Based on your config, I am assuming your CUCME phones are 3XXX. That strip pattern (taught by IPexpert) will take the last 4 digits of any inbound call. H323 has two legs. 1. Inbound Call - which reminds me... needs to be ^313...$ because Site A GK will send the tech-prefix to Site C Gateway (your output shows 31 as the tech prefix for Site C) 2. Outbound Call - now that you have accepted the call on dial peer 3000 (or whatever you decided to use) Site C Gateway will look to make another call out based on destination-pattern. Normally the call would be made to 313 but we will use the stip translation rule to make it 3XXX before trying to make the call Where is destination pattern 3XXX? You hidden CUCME dial-peers is where. Show voice dial-peer summary will show your hidden CUCME dial-peer which I am assuming have destination patter 3001 and 3002 Hope this helps. -Original Message- From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com] Sent: Sunday, February 17, 2013 6:16 PM To: Cory Gray Cc: 'ccie_voice' Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue Thank you so much for your efforts. I believe it may need a strip but i don't know exactly what or how to strip the prefix as with CUBE it works without need for translation rule. Thanks for info i will try and feed you back. Thanks, Hesham On Feb 17, 2013, at 4:08 PM, Cory Gray corygray22...@hotmail.com wrote: I am sorry. I had it backwards. I thought you had an issue routing to CUCM. For call into CUCME, you need this Dial peer voice 3000 voip Incoming called ^3...$ Dtmf-r h245a No vad Translation-profile in STRIP ! Voice translation-rule 1 Rule 1 /.+\(\)$/ /\1/ ! Voice translation-profile STRIP Translate called 1 -Original Message- From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com] Sent: Sunday, February 17, 2013 5:56 PM To: Cory Gray Cc: ccie_voice Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue Yes i am using g729 and i configured them from both sides CUCM side as region and location /devicepool and voice class codec as cme side. I am able to send calls from CME to CUCM but cucm unable to place calls to CME On Feb 17, 2013, at 3:51 PM, Cory Gray corygray22...@hotmail.com wrote: Should not have allow connections either unless you are doing cube but that should not break it. Debug h22r ans1 and look to see if there is detail on why the call is failing. Make sure you are using g729 as well Sent from my iPhone On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I did that and allow connections as well On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com wrote: Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - -
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
Yes thanks a lot I believe that's the whole issue of the prefix. That make sense and yes I believe you do understand what I am getting at totally and yes all what you've said are correct. I thank you so much for all your efforts. I will test it and feed you back but It may take with me a week or so to test but I have put it in my consideration. Many Thanks for all your efforts and it's highly appreciated. On Feb 17, 2013, at 4:25 PM, Cory Gray corygray22...@hotmail.com wrote: With CUBE, there is no tech prefix so that is why you don't need it here. Based on your config, I am assuming your CUCME phones are 3XXX. That strip pattern (taught by IPexpert) will take the last 4 digits of any inbound call. H323 has two legs. 1. Inbound Call - which reminds me... needs to be ^313...$ because Site A GK will send the tech-prefix to Site C Gateway (your output shows 31 as the tech prefix for Site C) 2. Outbound Call - now that you have accepted the call on dial peer 3000 (or whatever you decided to use) Site C Gateway will look to make another call out based on destination-pattern. Normally the call would be made to 313 but we will use the stip translation rule to make it 3XXX before trying to make the call Where is destination pattern 3XXX? You hidden CUCME dial-peers is where. Show voice dial-peer summary will show your hidden CUCME dial-peer which I am assuming have destination patter 3001 and 3002 Hope this helps. -Original Message- From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com] Sent: Sunday, February 17, 2013 6:16 PM To: Cory Gray Cc: 'ccie_voice' Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue Thank you so much for your efforts. I believe it may need a strip but i don't know exactly what or how to strip the prefix as with CUBE it works without need for translation rule. Thanks for info i will try and feed you back. Thanks, Hesham On Feb 17, 2013, at 4:08 PM, Cory Gray corygray22...@hotmail.com wrote: I am sorry. I had it backwards. I thought you had an issue routing to CUCM. For call into CUCME, you need this Dial peer voice 3000 voip Incoming called ^3...$ Dtmf-r h245a No vad Translation-profile in STRIP ! Voice translation-rule 1 Rule 1 /.+\(\)$/ /\1/ ! Voice translation-profile STRIP Translate called 1 -Original Message- From: Hesham Abdelkereem [mailto:heshamcentr...@gmail.com] Sent: Sunday, February 17, 2013 5:56 PM To: Cory Gray Cc: ccie_voice Subject: Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue Yes i am using g729 and i configured them from both sides CUCM side as region and location /devicepool and voice class codec as cme side. I am able to send calls from CME to CUCM but cucm unable to place calls to CME On Feb 17, 2013, at 3:51 PM, Cory Gray corygray22...@hotmail.com wrote: Should not have allow connections either unless you are doing cube but that should not break it. Debug h22r ans1 and look to see if there is detail on why the call is failing. Make sure you are using g729 as well Sent from my iPhone On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I did that and allow connections as well On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com wrote: Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
Since you have 2 zones i believe you must rely on zone prefix to determine which zone to select a gw from in order to route the call. In your config your zone prefix is 3... which seems incorrect by glancing at it. To route calls to CME via GK i would have a RP in CUCM like 4XXX and then prefix whatever the zone prefix is to it in the pattern. In your case prefix 31* to match your gateway registration to GK. Thus, my GK config would say zone prefix CUCME 31* The ARQ would come into GK with dialed digits of 31*4XXX , Then the gatekeeper would match tech prefix of 31*, and route to the gw registered in that zone (your CUCME). I would expect the call setup to arrive on CME with digits 31*4XXX and try to hit an inbound voip dialpeer, then you would need the inbound voip dialpeer to strip down to the last 4 digits, or 4XXX in this case, to match a registered ephone-dn. My inbound voip dialpeer on CME would only allow the g729 if my GK trunk was set to use g729. Apply a voice translation rule to the dialpeer to strip down to last 4 digits. If that ephone-dn is registered then it should ring. just my 2 cents... When you make the call from the CUCM phone, what output do you see on the CME with debug voip dialpeer? Do you see anything? On Sun, Feb 17, 2013 at 2:56 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Yes i am using g729 and i configured them from both sides CUCM side as region and location /devicepool and voice class codec as cme side. I am able to send calls from CME to CUCM but cucm unable to place calls to CME On Feb 17, 2013, at 3:51 PM, Cory Gray corygray22...@hotmail.com wrote: Should not have allow connections either unless you are doing cube but that should not break it. Debug h22r ans1 and look to see if there is detail on why the call is failing. Make sure you are using g729 as well Sent from my iPhone On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I did that and allow connections as well On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com wrote: Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 177.1.10.10 1720 177.1.10.10 32811 CUCM VOIP-GW H323-ID: CUCM_TRUNK_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 32788 CUCM VOIP-GW H323-ID: CUCM_TRUNK_2 Voice Capacity Max.= Avail.= Current.= 0 177.1.254.3 1720 177.1.254.3 63360 CUCM H323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CorpHQ(config-dial-peer)# CorpHQ(config-dial-peer)#do show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 31* Zone CUCM master gateway list: 177.1.254.3:1720 CUCME Prefix: 1*
Re: [OSL | CCIE_Voice] HQ-SC Gatekeeper Between CUCM and CME issue
I agree with you and and it does make sense. I have nothing now I just do that for my CCIE Voice lab preparation and I just try that during the rack rental. I have to do all that over again. As soon as I do it , I will let you know. I appreciate all your valuable information and thanks so much On Feb 17, 2013, at 5:18 PM, Steve Keller skeller...@gmail.com wrote: Since you have 2 zones i believe you must rely on zone prefix to determine which zone to select a gw from in order to route the call. In your config your zone prefix is 3... which seems incorrect by glancing at it. To route calls to CME via GK i would have a RP in CUCM like 4XXX and then prefix whatever the zone prefix is to it in the pattern. In your case prefix 31* to match your gateway registration to GK. Thus, my GK config would say zone prefix CUCME 31* The ARQ would come into GK with dialed digits of 31*4XXX , Then the gatekeeper would match tech prefix of 31*, and route to the gw registered in that zone (your CUCME). I would expect the call setup to arrive on CME with digits 31*4XXX and try to hit an inbound voip dialpeer, then you would need the inbound voip dialpeer to strip down to the last 4 digits, or 4XXX in this case, to match a registered ephone-dn. My inbound voip dialpeer on CME would only allow the g729 if my GK trunk was set to use g729. Apply a voice translation rule to the dialpeer to strip down to last 4 digits. If that ephone-dn is registered then it should ring. just my 2 cents... When you make the call from the CUCM phone, what output do you see on the CME with debug voip dialpeer? Do you see anything? On Sun, Feb 17, 2013 at 2:56 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Yes i am using g729 and i configured them from both sides CUCM side as region and location /devicepool and voice class codec as cme side. I am able to send calls from CME to CUCM but cucm unable to place calls to CME On Feb 17, 2013, at 3:51 PM, Cory Gray corygray22...@hotmail.com wrote: Should not have allow connections either unless you are doing cube but that should not break it. Debug h22r ans1 and look to see if there is detail on why the call is failing. Make sure you are using g729 as well Sent from my iPhone On Feb 17, 2013, at 5:39 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: I did that and allow connections as well On Feb 17, 2013, at 3:21 PM, Cory Gray corygray22...@hotmail.com wrote: Not sure if this is what is breaking it but you should not have voice class h323 1 on your ras dialpeer on site c Sent from my iPhone On Feb 17, 2013, at 4:59 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear All, I have tried to configure a gatekeeper between HQ-SC for interoperability between CME and HQ The issue is I am just able to call from CME to CUCM but Unable to call from CUCM to CME. Knowing that I have created a Device Pool , Route Pattern , Gatekeeper info , Gatekeeper controlled trunk for Gatekeeper to call CME from CUCM Side when I debug i always get ARJ Admission Rejection. I don't want to change anything in the technology prefix or anything. I don't want to use default technology prefix. I want show gatekeeper endpoints and show gatekeeper gw-type-prefix to be the same exactly. I just want to troubleshoot the issue of calling from CUCM to CME. Thank you so much for all your efforts However, here you are my configs GATEKEEPER HQ Router - SIDE voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip interface Loopback0 ip address 177.1.254.1 255.255.255.255 h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local CUCM cisco.com 177.1.254.1 zone local CUCME cisco.com zone prefix CUCM 1... zone prefix CUCM 2... zone prefix CUCME 3... gw-type-prefix 1* no shutdown SC Side interface Loopback0 ip address 177.1.254.3 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr 177.1.254.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 31 h323-gateway voip bind srcaddr 177.1.254.3 dial-peer voice 85 voip destination-pattern [12]...$ voice-class h323 1 session target ras dtmf-relay h245-alphanumeric CorpHQ(config-dial-peer)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 177.1.10.10 1720 177.1.10.10 32811 CUCM VOIP-GW H323-ID: CUCM_TRUNK_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.10.20 1720 177.1.10.20 32788 CUCM VOIP-GW H323-ID: CUCM_TRUNK_2 Voice Capacity Max.=
[OSL | CCIE_Voice] Directories button and Voicemail button
So I recently setup an IP Phone to use an External URL for directories. I am using the xml method on a web server to display the directories on the phone when pressed. On the phone configuration page i have set the directories URL to a file on my web server which gets the file and this works great to control the directories available, the order they are displayed, etc. However, now my voicemail button doesnt work on this phone. So i set the phones Services Provisioning setting to Both, which i think should use an external URL if specified and fall back to internal services if not. But this breaks my external URL and restores the voicemail button. In short it appears that i cannot manipulate the directory using external URL and maintain the voicemail button. It seems to be one or the other. Has anybody seen this before? It is reproduceable on my 7975 running 8-4-3S and my IP communicator. THis leads me to believe it is not a firmware issue but i might change out the firmware on the 7975s just for grins. My voicemail service remains an Enterprise Subscription and has not been modified for this task. thanks in advance! steve ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Qos big info
hi Guys ,On the switch how do I achieve this and where should I make these changes ?COS 5 should be in priority queueCOS 4, 6, 7 should be in Queue 2COS 3, 2, 3 should be in Queue 3COS 4, 0 should be in Queue 4Guarantee Queue 1 has the 25% of the bandwidth the other queues should share thebandwidth as 30 40 30.Once queue 2 reaches 60% capacity COS 4 packets should be droppedsinghGet Yourself a cool, short @in.com Email ID now! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Directories button and Voicemail button
Uncheck the enterprise services on your cucm this disables them internally, then set the services URL in enterprise parameters to the XML file you created and set all phones to both. This will allow you to do what you want. If you need to deny on or do something else then set that individual phone to its own file on the same server. Play with it, it is the best part of learning this. It will give a great sense of satisfaction when you can do whatever you want with this method. If you want more details on this look back to the discussion around 17 November 2012. Bill Sent from my iPad On Feb 17, 2013, at 7:40 PM, Steve Keller skeller...@gmail.com wrote: So I recently setup an IP Phone to use an External URL for directories. I am using the xml method on a web server to display the directories on the phone when pressed. On the phone configuration page i have set the directories URL to a file on my web server which gets the file and this works great to control the directories available, the order they are displayed, etc. However, now my voicemail button doesnt work on this phone. So i set the phones Services Provisioning setting to Both, which i think should use an external URL if specified and fall back to internal services if not. But this breaks my external URL and restores the voicemail button. In short it appears that i cannot manipulate the directory using external URL and maintain the voicemail button. It seems to be one or the other. Has anybody seen this before? It is reproduceable on my 7975 running 8-4-3S and my IP communicator. THis leads me to believe it is not a firmware issue but i might change out the firmware on the 7975s just for grins. My voicemail service remains an Enterprise Subscription and has not been modified for this task. thanks in advance! steve ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Custom Tones
I think Justin might be on to it but it has been a while since I have done this in the lab. Sent from my iPad On Feb 17, 2013, at 3:06 PM, Justin Carney justin.s.car...@gmail.com wrote: I haven't tested this recently, but it may help to make the join/leave tones use different frequencies, as well as using different time intervals for the cadence. I'm not sure why you're getting these strange results (two tones on join when your cadence only shows one and no tone on leave), but there may be some strange feature (or bug) that has to do with both join and leave using the same frequency. voice class custom-cptone leave dualtone conference frequency 300 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 cadence 800 -Justin On Sun, Feb 17, 2013 at 1:56 PM, William Bell b...@ucguerrilla.com wrote: I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910 translate calling 9101 translate called 910 ! voice translation-profile 911 translate calling 9111 translate called 911 ! voice translation-profile 97 translate
Re: [OSL | CCIE_Voice] Custom Tones
I'll give it a go tomorrow. I already reverted my pod this evening. I'll be doing another lab tomorrow, so I should be able to test this put by tomorrow afternoon. Sent from my iPad On Feb 17, 2013, at 9:14 PM, Bill whl...@gmail.com wrote: I think Justin might be on to it but it has been a while since I have done this in the lab. Sent from my iPad On Feb 17, 2013, at 3:06 PM, Justin Carney justin.s.car...@gmail.com wrote: I haven't tested this recently, but it may help to make the join/leave tones use different frequencies, as well as using different time intervals for the cadence. I'm not sure why you're getting these strange results (two tones on join when your cadence only shows one and no tone on leave), but there may be some strange feature (or bug) that has to do with both join and leave using the same frequency. voice class custom-cptone leave dualtone conference frequency 300 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 cadence 800 -Justin On Sun, Feb 17, 2013 at 1:56 PM, William Bell b...@ucguerrilla.com wrote: I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910
[OSL | CCIE_Voice] Call Recording with CUCM
Hi Folks, Which are the market options for call recording with CUCM beside of Zoom ? Thanks, -- Isamar Maia Cel. VIVO SSA: (55) 71-9146-8575 Cel. TIM SSA: (55) 71-9185-5264 Fixo: (55) 71-4062-8688 日本: +81-(0)3-4550-1212 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Directories button and Voicemail button
My requirement was to modify 1 phone to show Disabled when corporate directories was pressed, and to preserve the voicemail button. The piece that got it all working was to disable the native IP phone services for corp/personal/missed/place/received. After this step, setting the Services Provisioning to Both now allows my phone to look at the External URL supplied on the device level for my custom directory and allows the Voicemail button to workHowever this breaks the directories button for all other phones not using this URL. You must address this if your other phones are to remain unchanged. An example might be only a lobby phone needs corp directory disabled. To preserve the original directories behavior i had to create a second xml file for the other phones that listed everything in the default order. then make that the enterprise parameter URL. Finally set all phones Service Provisioning to both and reset all. Now all of my phones look at an external URL for directories. One phone is customized to display disabled when corp directory is pressed. the other phones find an xml file which preserves the original functionality when directories is pressed. hoepfully this helps someone out there.. , Steve Keller skeller...@gmail.com wrote: So I recently setup an IP Phone to use an External URL for directories. I am using the xml method on a web server to display the directories on the phone when pressed. On the phone configuration page i have set the directories URL to a file on my web server which gets the file and this works great to control the directories available, the order they are displayed, etc. However, now my voicemail button doesnt work on this phone. So i set the phones Services Provisioning setting to Both, which i think should use an external URL if specified and fall back to internal services if not. But this breaks my external URL and restores the voicemail button. In short it appears that i cannot manipulate the directory using external URL and maintain the voicemail button. It seems to be one or the other. Has anybody seen this before? It is reproduceable on my 7975 running 8-4-3S and my IP communicator. THis leads me to believe it is not a firmware issue but i might change out the firmware on the 7975s just for grins. My voicemail service remains an Enterprise Subscription and has not been modified for this task. thanks in advance! steve ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Custom Tones
I just had another idea...you are using the *dual* tone ( ie two tones/frequencies) command, but only specified one frequency. Try adding a second number on each frequency line. voice class custom-cptone leave dualtone conference frequency 300 350 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 750 cadence 800 If this works (using two tones on the frequency command lines) then my first idea of using different values may not apply but it could be useful to troubleshoot. On Feb 17, 2013 9:45 PM, Jason Lee jas7...@gmail.com wrote: I'll give it a go tomorrow. I already reverted my pod this evening. I'll be doing another lab tomorrow, so I should be able to test this put by tomorrow afternoon. Sent from my iPad On Feb 17, 2013, at 9:14 PM, Bill whl...@gmail.com wrote: I think Justin might be on to it but it has been a while since I have done this in the lab. Sent from my iPad On Feb 17, 2013, at 3:06 PM, Justin Carney justin.s.car...@gmail.com wrote: I haven't tested this recently, but it may help to make the join/leave tones use different frequencies, as well as using different time intervals for the cadence. I'm not sure why you're getting these strange results (two tones on join when your cadence only shows one and no tone on leave), but there may be some strange feature (or bug) that has to do with both join and leave using the same frequency. voice class custom-cptone leave dualtone conference frequency 300 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 cadence 800 -Justin On Sun, Feb 17, 2013 at 1:56 PM, William Bell b...@ucguerrilla.comwrote: I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any
