[OSL | CCIE_Voice] I Passed!!!!!!!!!! Certified (CCIE#26063)
Hello All, I had my exams in SAN JOSE yesterday and got the result this morning It was my second attempt Thanks all for answering all questions i posted and most especially IP Expert, i attended the bootcamp two weeks ago. Amy is a wonderful instructor during and after the training. Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] I Passed!!!!!!!!!! Certified (CCIE#26063)
No OEQ but lots of troubleshooting IPExpert workbooks is the way to go On Tue, May 11, 2010 at 5:37 PM, KatGuru gkr2...@yahoo.com wrote: Congrats !!! Please let us know if you had the OEQs in the lab. Thank you. --- On *Wed, 5/12/10, Omotayo adefilabi...@gmail.com* wrote: From: Omotayo adefilabi...@gmail.com Subject: [OSL | CCIE_Voice] I Passed!! Certified (CCIE#26063) To: OSL Group ccie_voice@onlinestudylist.com Date: Wednesday, May 12, 2010, 8:24 AM Hello All, I had my exams in SAN JOSE yesterday and got the result this morning It was my second attempt Thanks all for answering all questions i posted and most especially IP Expert, i attended the bootcamp two weeks ago. Amy is a wonderful instructor during and after the training. Regards -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Frame relay traffic shaping
Hello all, If questions does not explicitly say we should configure shaping. Do we have to on the physical interface? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Frame relay traffic shaping
thanks On Sun, May 9, 2010 at 7:33 PM, bkvalent...@gmail.com bkvalent...@gmail.com wrote: I wouldn't waste time configuring anything that won't earn you points. The test isn't about how closely you follow best practices. Do what is asked. If you aren't sure what is being asked, check with the proctor. Brian - Reply message - From: Omotayo adefilabi...@gmail.com Date: Sun, May 9, 2010 9:09 pm Subject: [OSL | CCIE_Voice] Frame relay traffic shaping To: OSL Group ccie_voice@onlinestudylist.com Hello all, If questions does not explicitly say we should configure shaping. Do we have to on the physical interface? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Calls rejected by h323 gateway
Hello, bri router has been configured as h323 gateway. call from the pstn to br1 phone are rejected with the following debug but calls to pstn phone are completed BR1-RTR# BR1-RTR# May 9 02:27:17.246: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0085 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Calling Party Number i = 0x4180, '8632683' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0xA1, '6178631002' Plan:ISDN, Type:National May 9 02:27:17.250: ISDN Se0/0/0:23 Q931: Received SETUP callref = 0x8085 callID = 0x0005 switch = primary-ni interface = User May 9 02:27:17.250: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=8632683, Called Number=1002, Voice-Interface=0x49CFCF98, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.254: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) a BR1-RTR#fter DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=100 May 9 02:27:17.254: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=8632683, Called Number=1002, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_FAX May 9 02:27:17.254: //-1//DPM/dpAssociateIncomingPeerCore: Result=NO_MATCH(-1) After All Match Rules Attempt May 9 02:27:17.254: //-1//DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.254: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 May 9 02:27:17.254: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST May 9 02:27:17.254: //-1//DPM/dpMatchPeers: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=200 2: Dial-peer Tag=300 May 9 02:27:17.266: //-1/38905CC58006/DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.266: //-1/38905CC58006/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 May 9 02:27:17.266: //-1/38905CC58006/DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST May 9 02:27:17.266: //-1/38905CC58006/DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=200 2: Dial-peer Tag=300 May 9 02:27:17.266: //-1//DPM/dpMatchPeersCore: Calling Number=1002, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.266: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 May 9 02:27:17.266: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST May 9 02:27:17.270: //-1//DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=200 2: Dial-peer Tag=300 May 9 02:27:17.270: //-1//DPM/dpMatchPeersCore: Calling Number=1002, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.270: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 May 9 02:27:17.270: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST May 9 02:27:17.270: //-1//DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=200 2: Dial-peer Tag=300 May 9 02:27:17.270: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=1002, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.270: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=200 May 9 02:27:17.270: //-1//DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.270: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 May 9 02:27:17.270: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST May 9 02:27:17.270: //-1//DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=200 2: Dial-peer Tag=300 May 9 02:27:17.274: //-1/38905CC58006/DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.274: //-1/38905CC58006/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 May
Re: [OSL | CCIE_Voice] Calls rejected by h323 gateway
=DP_MATCH_DEST; Called Number=1002 *May 9 04:11:17.937: //-1/C04AA4488004/DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST *May 9 04:11:17.937: //-1/C04AA4488004/DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=200 2: Dial-peer Tag=300 *May 9 04:11:17.941: ISDN Se0/0/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x809E Channel ID i = 0xA98381 Exclusive, Channel 1 *May 9 04:11:18.237: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x809E Cause i = 0x809B - Destination out of order *May 9 04:11:18.249: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x009E *May 9 04:11:18.249: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x809E BR1-RTR# BR1-RTR# On Sat, May 8, 2010 at 4:32 PM, Vik Malhi vma...@ipexpert.com wrote: Have I missed something or are you lacking a h323-g voip bind srcaddr -- Vik Malhi – CCIE #13890 Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On May 8, 2010, at 3:31 PM, Omotayo adefilabi...@gmail.com wrote: Hello, bri router has been configured as h323 gateway. call from the pstn to br1 phone are rejected with the following debug but calls to pstn phone are completed BR1-RTR# BR1-RTR# May 9 02:27:17.246: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0085 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Calling Party Number i = 0x4180, '8632683' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0xA1, '6178631002' Plan:ISDN, Type:National May 9 02:27:17.250: ISDN Se0/0/0:23 Q931: Received SETUP callref = 0x8085 callID = 0x0005 switch = primary-ni interface = User May 9 02:27:17.250: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=8632683, Called Number=1002, Voice-Interface=0x49CFCF98, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.254: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) a BR1-RTR#fter DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=100 May 9 02:27:17.254: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=8632683, Called Number=1002, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_FAX May 9 02:27:17.254: //-1//DPM/dpAssociateIncomingPeerCore: Result=NO_MATCH(-1) After All Match Rules Attempt May 9 02:27:17.254: //-1//DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.254: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 May 9 02:27:17.254: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST May 9 02:27:17.254: //-1//DPM/dpMatchPeers: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=200 2: Dial-peer Tag=300 May 9 02:27:17.266: //-1/38905CC58006/DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.266: //-1/38905CC58006/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 May 9 02:27:17.266: //-1/38905CC58006/DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST May 9 02:27:17.266: //-1/38905CC58006/DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=200 2: Dial-peer Tag=300 May 9 02:27:17.266: //-1//DPM/dpMatchPeersCore: Calling Number=1002, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH May 9 02:27:17.266: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 May 9 02:27:17.266: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST May 9 02:27:17.270: //-1//DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched
Re: [OSL | CCIE_Voice] Calls rejected by h323 gateway
Hello, Still disconnecting for br1 phone 2. When voicemail is configured for br1 phone 2. it goes to his voicemail but if other pstn phones calls br1 phone 2 it connects going through same gateway On Sat, May 8, 2010 at 5:47 PM, Vik Malhi vma...@ipexpert.com wrote: Change the ucm service parameter to be trunk/gw CSS as opposed to rdp CSS and report back. -- Vik Malhi – CCIE #13890 Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On May 8, 2010, at 5:26 PM, Omotayo adefilabi...@gmail.com wrote: Hello, Dont know why its behaving this way; i can call other phones in br1(1001) i can call the mva (1003) and make an outbound call but i cant call 1002 note, the pstn phone 2 (8632683) is the remote destination profile for extension 1002- Could this be the reason Thanks On Sat, May 8, 2010 at 5:18 PM, Vik Malhi vma...@ipexpert.com wrote: Is the device name of your h323 gw 10.10.201.1. Aldo reset it. Check that CSS of gw can see phone pt (and nothing else). Check location of gw and phone. Check other phones (internal) can call the phone. -- Vik Malhi – CCIE #13890 Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.comvma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: http://www.ipexpert.com/chat www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On May 8, 2010, at 5:12 PM, Omotayo adefilabi...@gmail.com wrote: Hello, I have added it interface Vlan240 ip address 10.10.201.1 255.255.255.0 ip helper-address 10.10.210.10 h323-gateway voip bind srcaddr 10.10.201.1 Calls are still failing *May 9 04:11:17.913: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x009E Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Calling Party Number i = 0x4180, '8632683' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0xA1, '6178631002' Plan:ISDN, Type:National *May 9 04:11:17.917: ISDN Se0/0/0:23 Q931: Received SETUP callref = 0x809E callID = 0x0003 switch = primary-ni interface = User *May 9 04:11:17.917: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=8632683, Called Number=1002, Voice-Interface=0x49CC2340, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *May 9 04:11:17.917: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success( BR1-RTR#0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=100 *May 9 04:11:17.917: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=8632683, Called Number=1002, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_FAX *May 9 04:11:17.921: //-1//DPM/dpAssociateIncomingPeerCore: Result=NO_MATCH(-1) After All Match Rules Attempt *May 9 04:11:17.921: //-1//DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH *May 9 04:11:17.921: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 *May 9 04:11:17.921: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST *May 9 04:11:17.921: //-1//DPM/dpMatchPeers: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=200 2: Dial-peer Tag=300 *May 9 04:11:17.929: //-1/C04AA4488004/DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH *May 9 04:11:17.929
[OSL | CCIE_Voice] lab8 vol 2- PSTN PHONE NOT REGISTERING
Hello, My PSTN phone for lab 8 vol 2 is not registering Any one with similar issue ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Music on hold on h323
Hello All, Never mind Its working now Rgd On Thu, May 6, 2010 at 5:25 PM, Omotayo adefilabi...@gmail.com wrote: Hello, I have br1 registered as h323 gateway with music hold configured the MOH is assigned to br1 phones When br1 phone put pstn phone on hold the multicast is used but i dont hear the music Anyone with an idea of what th eissue might be Perf class (Cisco MOH Device) has instances and values: MOH_2 - MOHHighestActiveResources = 1 MOH_2 - MOHMulticastResourceActive = 1 MOH_2 - MOHMulticastResourceAvailable = 24 MOH_2 - MOHOutOfResources = 0 MOH_2 - MOHTotalMulticastResources = 25 MOH_2 - MOHTotalUnicastResources = 250 MOH_2 - MOHUnicastResourceActive = 0 MOH_2 - MOHUnicastResourceAvailable= 250 MOH_3 - MOHHighestActiveResources = 1 MOH_3 - MOHMulticastResourceActive = 0 MOH_3 - MOHMulticastResourceAvailable = 25 MOH_3 - MOHOutOfResources = 0 MOH_3 - MOHTotalMulticastResources = 25 MOH_3 - MOHTotalUnicastResources = 250 MOH_3 - MOHUnicastResourceActive = 0 MOH_3 - MOHUnicastResourceAvailable= 250 thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA Call Disconnect
Hello, The calling number is recognized, that is the reason it prompt for PIN and not remote destination number Amy, i have checked the CSS, the CSS of the remote destianation profile is CSS-MVA There is a route pattern 9.011! with pt-MVA It still disconnecting Thanks On Thu, May 6, 2010 at 5:56 PM, Jeff Cotter jcot...@voxns.com wrote: Try checking the calling line ID of the device you are using to initiate the call matches a remote destination profile that is associated with an internal DN. If call manager does not recognize the caller it will drop the call. HTH, Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Thursday, May 06, 2010 5:22 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 51, Issue 39 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Switch QoS - Vol.2 Lab 8 Q5.3 (Sergio Polizer) 2. Re: Switch QoS - Vol.2 Lab 8 Q5.3 (Mad Kiwi) 3. Cisco 837 for VPN to Proctors Lab (kerboute kerboute) 4. Lab 5c MVA (vccie2010) 5. unable to register IP Communicator to CME (vccie2010) 6. MVA Call Disconnect (Omotayo) 7. Re: Agent-Based Routing (Amy Ryan) -- Message: 1 Date: Thu, 6 May 2010 18:32:37 -0300 From: Sergio Polizer spoli...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Switch QoS - Vol.2 Lab 8 Q5.3 To: ccievoi...@gmail.com, ccie_voice@onlinestudylist.com Message-ID: bay114-w45e890f55bcd127e3d17e4cb...@phx.gbl Content-Type: text/plain; charset=iso-8859-1 Hi, This is the expected behavior. According to Catalyst 3750 Switch Software Configuration Guide, 12.2(44)SE: Do not use the show policy-map interface privileged EXEC command to display classification information for incoming traffic. The control-plane and interface keywords are not supported, and the statistics shown in the display should be ignored. HTH, Sergio. Date: Thu, 6 May 2010 21:40:17 +1000 From: ccievoi...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Switch QoS - Vol.2 Lab 8 Q5.3 Hi everyone, I'm working through the HQ (3750) Switch QOS lab, where we use AutoQos, should I be able to see packet classification under the show policy-map interface statement? As far as I can tell straight out of the box Auto QoS isn't matching the packets. Any thoughts? -- Cheers, kiwi _ QUEM VOC? QUER SER HOJE NO MESSENGER? TRANSFORME SUA FOTO, ? GR?TIS. http://ilm.windowslive.com.br/?ocid=ILM:ILM:Hotmail:Tagline:1x1:Tagline -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100506/420c3ad1/attachment-0001.htm -- Message: 2 Date: Fri, 7 May 2010 07:49:20 +1000 From: Mad Kiwi ccievoi...@gmail.com Subject: Re: [OSL | CCIE_Voice] Switch QoS - Vol.2 Lab 8 Q5.3 To: Sergio Polizer spoli...@hotmail.com Cc: ccie_voice@onlinestudylist.com Message-ID: z2zcb331f6e1005061449kb0d1dbccv78e35d32a019b...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Thanks Sergio, That's what is was struggling to find last night. Regards Kiwi On Fri, May 7, 2010 at 7:32 AM, Sergio Polizer spoli...@hotmail.com wrote: Hi, This is the expected behavior. According to Catalyst 3750 Switch Software Configuration Guide, 12.2(44)SE: Do not use the* show policy-map* *interface* privileged EXEC command to display classification information for incoming traffic. The * control-plane* and *interface* keywords are not supported, and the statistics shown in the display should be ignored. HTH, Sergio. -- Date: Thu, 6 May 2010 21:40:17 +1000 From: ccievoi...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Switch QoS - Vol.2 Lab 8 Q5.3 Hi everyone, I'm working through the HQ (3750) Switch QOS lab, where we use AutoQos, should I be able to see packet classification under the show policy-map interface statement? As far as I can tell straight out of the box Auto QoS isn't matching the packets. Any thoughts? -- Cheers, kiwi -- EM 2009 ACONTECERAM 250.362 FRAUDES NA INTERNET. CLIQUE PARA LER
Re: [OSL | CCIE_Voice] Music on hold on h323
hello, i bounced the config and applied back and it worked Rgd On Thu, May 6, 2010 at 6:22 PM, vccie2010 vccie2...@gmail.com wrote: Will appreciate if you may please give ur to cents on what was the issue for the benefit... On Thu, May 6, 2010 at 5:25 PM, Omotayo adefilabi...@gmail.com wrote: Hello, I have br1 registered as h323 gateway with music hold configured the MOH is assigned to br1 phones When br1 phone put pstn phone on hold the multicast is used but i dont hear the music Anyone with an idea of what th eissue might be Perf class (Cisco MOH Device) has instances and values: MOH_2 - MOHHighestActiveResources = 1 MOH_2 - MOHMulticastResourceActive = 1 MOH_2 - MOHMulticastResourceAvailable = 24 MOH_2 - MOHOutOfResources = 0 MOH_2 - MOHTotalMulticastResources = 25 MOH_2 - MOHTotalUnicastResources = 250 MOH_2 - MOHUnicastResourceActive = 0 MOH_2 - MOHUnicastResourceAvailable= 250 MOH_3 - MOHHighestActiveResources = 1 MOH_3 - MOHMulticastResourceActive = 0 MOH_3 - MOHMulticastResourceAvailable = 25 MOH_3 - MOHOutOfResources = 0 MOH_3 - MOHTotalMulticastResources = 25 MOH_3 - MOHTotalUnicastResources = 250 MOH_3 - MOHUnicastResourceActive = 0 MOH_3 - MOHUnicastResourceAvailable= 250 thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Destination out of order
Hello All, working on Lab 7 volume 2 When hq pstn number call hq phones i get the following messages Any one with a fix NB: Call from other pstn number to hq phones works well HQ-RTR(config)# May 5 01:36:13.476: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x8095 Cause i = 0x829B - Destination out of order May 5 01:36:13.492: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x0095 May 5 01:36:13.524: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x8095 HQ-RTR(config)# May 5 01:41:19.047: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0096 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Calling Party Number i = 0x4180, '7773434' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0xA1, '4087775002' Plan:ISDN, Type:National May 5 01:41:19.147: ISDN Se0/0/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x8096 Channel ID i = 0xA98381 Exclusive, Channel 1 HQ-RTR(config)# May 5 01:41:19.147: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x8096 Cause i = 0x829B - Destination out of order May 5 01:41:19.163: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x0096 May 5 01:41:19.191: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x8096 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] utils ntp status
Hello, Does any one think there is something wrong with the PTSN Router. UTC time is showing 22:13:23 UTC 2010 instead of 18:20 a the time of posting On the hq router, it shows HQ-RTR#sh clock 15:21:33.411 pst Mon May 3 2010 HQ-RTR# while on hq phone it shows 18:22 22/03/2010 hq is pst -8 Any know why i have these discrepancies? admin:utils ntp status ntpd (pid 4973) is running... remote refid st t when poll reach delay offset jitter == 127.127.1.0 LOCAL(0)10 l4 64 3770.0000.000 0.004 *10.10.110.1 10.10.100.2 5 u 77 128 3771.200 -15.310 6.770 synchronised to NTP server (10.10.110.1) at stratum 6 time correct to within 35 ms polling server every 128 s Current time in UTC is : Mon May 3 22:13:23 UTC 2010 Current time in America/New_York is : Mon May 3 18:13:23 EDT 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] login into CME via the web
Hello, I usually have issue logging into the CUE via GUI. When i supply the ip add of the loopback and the username and password entered under the telephony-service i get Loin into callmanger as administrator failed When i use vlan 400, i get same result below is my config voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip bind control source-interface Vlan400 bind media source-interface Vlan400 ! ip http server ip http authentication local no ip http secure-server ip http path FLASH:/GUI telephony-service sdspfarm units 2 sdspfarm transcode sessions 2 sdspfarm tag 1 Sitec-conf no privacy conference hardware no auto-reg-ephone max-ephones 5 max-dn 10 ip source-address 10.10.202.1 port 2000 time-zone 42 voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Apr 21 2010 08:41:40 ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] login into CME via the web
Hello, i can log in now. voice mail works but mwi is not turned on i am using Outcalling What do you think am missing out thanks On Wed, Apr 21, 2010 at 2:59 AM, Amy Ryan ar...@ipexpert.com wrote: It is possible you may have the wrong type of license assigned to the CUE module depending on what type of CUE integration you are performing. You can do “show software licenses” to determine the application mode being used. If you are using Proctor Labs, there is a FTP server running on UCCX for this purpose. se-10-10-202-2# sh software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCM - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 2 - Max enabled languages: 2 Let us know if this helps out. Amy --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: *ar...@ipexpert.com *Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat * http://www.ipexpert.com/chat* eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities *http://www.ipexpert.com/communities* and our public website at www.ipexpert.com *http://www.ipexpert.com/* -- *From: *Omotayo adefilabi...@gmail.com *Date: *Wed, 21 Apr 2010 01:52:16 +0100 *To: *OSL Group ccie_voice@onlinestudylist.com *Subject: *[OSL | CCIE_Voice] login into CME via the web Hello, I usually have issue logging into the CUE via GUI. When i supply the ip add of the loopback and the username and password entered under the telephony-service i get Loin into callmanger as administrator failed When i use vlan 400, i get same result below is my config voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip bind control source-interface Vlan400 bind media source-interface Vlan400 ! ip http server ip http authentication local no ip http secure-server ip http path FLASH:/GUI telephony-service sdspfarm units 2 sdspfarm transcode sessions 2 sdspfarm tag 1 Sitec-conf no privacy conference hardware no auto-reg-ephone max-ephones 5 max-dn 10 ip source-address 10.10.202.1 port 2000 time-zone 42 voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Apr 21 2010 08:41:40 ! -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Forward Unregistered in SRST
Hello, i have configured SRST FOR Br1 so that hq and br2 phones can call it when unregistered from UCM( in SRST). i have the CSS as CSS-FUR with partition pt-fur i created a route pattern \+.1617! in pt-fur i have the number +16178631002 assigned whenn unregistered When i enable the SRST, it shows unknown on the callmanager, so when hq dial 1002. it goes to voicemail instead of ringing Any idea on why its showing Unknown instead of Unregistered Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Forward Unregistered in SRST
Hello, i didnt uncheck it. i guess that might be the problem Thanks, will check and let you have the feedback Regards On Tue, Apr 13, 2010 at 11:36 AM, o...@ipexpert.com wrote: Hello, Make sure the vm checkbox is not checked for the cfur internal line configuration this takes precedence over the cfur specific configuration, -Original Message- From: Omotayo adefilabi...@gmail.com Date: Tue, 13 Apr 2010 11:04:59 To: OSL Groupccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Forward Unregistered in SRST ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 - MWI
am using g711 because of the GK with CUBE qquestion On Sat, Apr 10, 2010 at 12:34 AM, Ashar Siddiqui siddas...@gmail.comwrote: Are you usig voice class codec in default Incoming voip dial-peer? Put only g729r8 in incoming voip dial-peer and remove voice class codec. Ash On 09/04/2010 13:18, Amy Ryan wrote: Omotayo, When integrating CUCME and CUE it is best to use sip-notify vs. rtp-nte for dtmf configured on the dial-peer. dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] * dtmf-relay sip-notify * codec g711ulaw no vad The dtmf-relay rtp-nte command sets the SIP DTMF relay mechanism to use the media path to relay incoming and outgoing DTMF signals to Cisco Unity Express. The dtmf-relay sip-notify command sets the SIP DTMF relay mechanism to use Unsolicited-Notify messages to relay incoming and outgoing DTMF signals. HTH, Amy --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: *ar...@ipexpert.com *Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat * http://www.ipexpert.com/chat* eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities *http://www.ipexpert.com/communities* and our public website at www.ipexpert.com *http://www.ipexpert.com/* *From: *Omotayo adefilabi...@gmail.com *Date: *Fri, 9 Apr 2010 09:40:36 +0100 *To: *Roger Källberg roger.kallb...@cygate.se *Cc: *OSL Group ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] Lab 1 - MWI dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay rtp-nte codec g711ulaw no vad ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Thanks, Ashar Siddiqui ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 - MWI
i cant not use g729r8 because the call from hq to br2 phone requirement says; hq -cube = 16k cube-br2 = 128k So i used g711ulaw for the incoming dial-peer Does all these affect MWI not turning on ??? On Sat, Apr 10, 2010 at 6:19 PM, Ashar Siddiqui siddas...@gmail.com wrote: Remove voice class codec from Incoming voip dial-peer and hardcode G729r8. Ash On 10/04/2010 16:40, Omotayo wrote: am using g711 because of the GK with CUBE qquestion On Sat, Apr 10, 2010 at 12:34 AM, Ashar Siddiqui siddas...@gmail.comwrote: Are you usig voice class codec in default Incoming voip dial-peer? Put only g729r8 in incoming voip dial-peer and remove voice class codec. Ash On 09/04/2010 13:18, Amy Ryan wrote: Omotayo, When integrating CUCME and CUE it is best to use sip-notify vs. rtp-nte for dtmf configured on the dial-peer. dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] * dtmf-relay sip-notify * codec g711ulaw no vad The dtmf-relay rtp-nte command sets the SIP DTMF relay mechanism to use the media path to relay incoming and outgoing DTMF signals to Cisco Unity Express. The dtmf-relay sip-notify command sets the SIP DTMF relay mechanism to use Unsolicited-Notify messages to relay incoming and outgoing DTMF signals. HTH, Amy --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: *ar...@ipexpert.com *Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat * http://www.ipexpert.com/chat* eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities *http://www.ipexpert.com/communities* and our public website at www.ipexpert.com *http://www.ipexpert.com/* *From: *Omotayo adefilabi...@gmail.com *Date: *Fri, 9 Apr 2010 09:40:36 +0100 *To: *Roger Källberg roger.kallb...@cygate.se *Cc: *OSL Group ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] Lab 1 - MWI dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay rtp-nte codec g711ulaw no vad ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Thanks, Ashar Siddiqui -- Thanks, Ashar Siddiqui ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab 1 - MWI
Hello, I am working on volume 2 lab 1, when hq phone calls br2 phone and hq phone leave a message on br2 phone, it gets to the voicemail but the MWI does not turn. Any one with an idea of what could be the problem. Below is the relevant config dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay rtp-nte codec g711ulaw no vad ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 2 sdspfarm tag 1 br2-xcoder no auto-reg-ephone authentication credential admin cisco max-ephones 10 max-dn 10 ip source-address 10.10.110.3 port 2000 url services http://10.10.202.2/voiceview/common/login.do url authentication http://10.10.202.1/CCMCIP/authenticate.asp voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T web admin system name admin password cisco transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Apr 08 2010 23:30:55 ephone-dn 1 dual-line number 3001 no-reg primary label Br2 Phn1 name Br2 Phn1 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 2 dual-line number 3002 no-reg primary label Br2 Phn2 name Br2 Phn2 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 3 number 3999 no-reg primary mwi on ! ! ephone-dn 4 number 3998 no-reg primary mwi off ! ! ephone 1 device-security-mode none mac-address 001E.0B2D.F37D username Br2Phn1 password cisco type CIPC button 1:1 ! ! ! ephone 2 device-security-mode none mac-address 001E.EC15.996D username Br2Phn2 password cisco type CIPC button 1:2 ! thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 - MWI
Hello, the CUE uses the configured MWI -outdialling I will check the debug to see if its been used Regards 2010/4/9 Roger Källberg roger.kallb...@cygate.se Hi, Have you checked with debub voip dialpeer that the CUE dials your MWI on/off numbers? There is a bug that sometimes makes it use the default MWI extensions. I believe that they are …. and 8889…. If so change the MWI settings temporary in CUE to not include outdial, then do a resync of MWI and look at the debug, you should not get any output. Then change it back to outdial and resync once more, this time you should get output in the debug. Check the debug to see that CUE now uses your MWI numbers. *Roger Källberg* Unified Communication Consultant Cygate AB *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* den 9 april 2010 09:01 *To:* OSL Group *Subject:* [OSL | CCIE_Voice] Lab 1 - MWI Hello, I am working on volume 2 lab 1, when hq phone calls br2 phone and hq phone leave a message on br2 phone, it gets to the voicemail but the MWI does not turn. Any one with an idea of what could be the problem. Below is the relevant config dial-peer voice 3160 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 399[89] dtmf-relay rtp-nte codec g711ulaw no vad ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 2 sdspfarm tag 1 br2-xcoder no auto-reg-ephone authentication credential admin cisco max-ephones 10 max-dn 10 ip source-address 10.10.110.3 port 2000 url services http://10.10.202.2/voiceview/common/login.do url authentication http://10.10.202.1/CCMCIP/authenticate.asp voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T web admin system name admin password cisco transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Apr 08 2010 23:30:55 ephone-dn 1 dual-line number 3001 no-reg primary label Br2 Phn1 name Br2 Phn1 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 2 dual-line number 3002 no-reg primary label Br2 Phn2 name Br2 Phn2 call-forward busy 3600 call-forward noan 3600 timeout 10 ! ! ephone-dn 3 number 3999 no-reg primary mwi on ! ! ephone-dn 4 number 3998 no-reg primary mwi off ! ! ephone 1 device-security-mode none mac-address 001E.0B2D.F37D username Br2Phn1 password cisco type CIPC button 1:1 ! ! ! ephone 2 device-security-mode none mac-address 001E.EC15.996D username Br2Phn2 password cisco type CIPC button 1:2 ! thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 - MWI
thanks, i will check all this and have a feedback On Fri, Apr 9, 2010 at 5:36 PM, Mike Peterson polobi...@yahoo.com wrote: Omotayo, Amy you are right about dtmf-relay sip-notify but I think he also has something ls which doesn't match. The best practice is that always you need to check your configs at both ends in this case CME and CUE . So could you post the relevant MWI configs from CUE too ? Without knowing your relevant MWI config from CUE , I think that the DN (incoming called-number) for MWI in CUE doesn't match the CME MWI. Take a look at my sample bellow from a real world project about 2 weeks ago on CME/CUE: CME: dial-peer voice 1001 voip description MWI Inbound Dial-peer destination-pattern ^100[12]$ session protocol sipv2 session target ipv4:172.16.64.20 incoming called-number 100[12] dtmf-relay sip-notify codec g711ulaw no vad CUE: ccn trigger sip phonenumber 1000 application voicemail enabled maxsessions 8 end trigger .. voicemail broadcast mwi voicemail callerid voicemail default mailboxsize 10600 voicemail broadcast recording time 300 voicemail default messagesize 280 Summary , in this case incoming called-number 100[12] match CUE VM Extn 1000. hth ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calls from h323 gateway rejected by call manager
You need to add dial-peer voice 100 pots incoming called-number . direct-inward-dial On Wed, Apr 7, 2010 at 7:12 AM, sean hurricane shurric...@gmail.com wrote: I have an h323 gateway provisioned in call manageri can successfully make outbound calls but inbound PSTN calls are not ringing the phone (translation pattern is configured to strip inbound calls from 10 to 4) i am getting the following error in the trace file 04/06/2010 23:57:43.327 CCM|H225Handler::SdlConnectionInd, rejecting the TCP connection from IP=10.10.201.1. Incoming H225 call on Port=1720, can't find this port in the registered H323 endpoints port list. |CLID::StandAloneClusterNID::10.10.210.11CT::0,0,0,0.0IP::DEV::LVL::ErrorMASK::0100 04/06/2010 23:57:59.551 CCM|H225Handler::SdlConnectionInd, rejecting the TCP connection from IP=10.10.201.1. Incoming H225 call on Port=1720, can't find this port in the registered H323 endpoints port list. |CLID::StandAloneClusterNID::10.10.210.11CT::0,0,0,0.0IP::DEV::LVL::ErrorMASK::0100 04/06/2010 23:58:51.835 CCM|H225Handler::SdlConnectionInd, rejecting the TCP connection from IP=10.10.201.1. Incoming H225 call on Port=1720, can't find this port in the registered H323 endpoints port list. |CLID::StandAloneClusterNID::10.10.210.11CT::0,0,0,0.0IP::DEV::LVL::ErrorMASK::0100 *Router Config * voice translation-rule 1 rule 1 /415888\(1...\)/ /\1/ voice translation-profile INBOUND translate called 1* *voice-port 0/2/1:23 translation-profile incoming INBOUND* *dial-peer voice 3000 voip destination-pattern 1...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.11 dtmf-relay h245-alphanumeric no vad dial-peer voice 3005 voip preference 1 destination-pattern 1...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric no vad BR1#sh run int fa0/0.240 Building configuration... Current configuration : 204 bytes ! interface FastEthernet0/0.240 encapsulation dot1Q 240 ip address 10.10.201.1 255.255.255.0 ip helper-address 10.10.200.3 h323-gateway voip interface h323-gateway voip bind srcaddr 10.10.201.1 Weird thing is i can use csim start to call phones and they ring but inbound PSTN calls gets the error msg above any thots? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calls from h323 gateway rejected by call manager
Sean, the last config you posted lloks like its on the br2 router. you should have the dial-peer voice 1 pots incoming called-number . direct-inward-dial on the br1 router On Wed, Apr 7, 2010 at 7:29 AM, sean hurricane shurric...@gmail.com wrote: i had that in another dial-peer, see below: dial-peer voice 1 pots incoming called-number . direct-inward-dial dial-peer voice 1000 voip destination-pattern 3...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.11 dtmf-relay h245-alphanumeric no vad dial-peer voice 1005 voip preference 1 destination-pattern 3...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric no vad dial-peer voice 3600 voip destination-pattern 3600$ session protocol sipv2 session target ipv4:10.10.202.2 dtmf-relay sip-notify codec g711ulaw no vad dial-peer voice 3 pots translation-profile outgoing OUTBOUND destination-pattern 999 port 0/2/0:15 forward-digits 3 dial-peer voice 5 pots translation-profile outgoing OUTBOUND destination-pattern 9%112 port 0/2/0:15 forward-digits 3 dial-peer voice 7 pots translation-profile outgoing LOCAL destination-pattern 97... port 0/2/0:15 forward-digits 8 dial-peer voice 11 pots translation-profile outgoing +DIAL destination-pattern 90T port 0/2/0:15 forward-digits 12 dial-peer voice 666 voip service clid_authen_collect destination-pattern 907976852817 session target ipv4:10.10.110.3 incoming called-number 907976852817 dtmf-relay h245-alphanumeric codec g711ulaw no vad dial-peer voice 9 pots translation-profile incoming MVA service mva incoming called-number 3777 no digit-strip direct-inward-dial On Wed, Apr 7, 2010 at 2:23 AM, Omotayo adefilabi...@gmail.com wrote: You need to add dial-peer voice 100 pots incoming called-number . direct-inward-dial On Wed, Apr 7, 2010 at 7:12 AM, sean hurricane shurric...@gmail.comwrote: I have an h323 gateway provisioned in call manageri can successfully make outbound calls but inbound PSTN calls are not ringing the phone (translation pattern is configured to strip inbound calls from 10 to 4) i am getting the following error in the trace file 04/06/2010 23:57:43.327 CCM|H225Handler::SdlConnectionInd, rejecting the TCP connection from IP=10.10.201.1. Incoming H225 call on Port=1720, can't find this port in the registered H323 endpoints port list. |CLID::StandAloneClusterNID::10.10.210.11CT::0,0,0,0.0IP::DEV::LVL::ErrorMASK::0100 04/06/2010 23:57:59.551 CCM|H225Handler::SdlConnectionInd, rejecting the TCP connection from IP=10.10.201.1. Incoming H225 call on Port=1720, can't find this port in the registered H323 endpoints port list. |CLID::StandAloneClusterNID::10.10.210.11CT::0,0,0,0.0IP::DEV::LVL::ErrorMASK::0100 04/06/2010 23:58:51.835 CCM|H225Handler::SdlConnectionInd, rejecting the TCP connection from IP=10.10.201.1. Incoming H225 call on Port=1720, can't find this port in the registered H323 endpoints port list. |CLID::StandAloneClusterNID::10.10.210.11CT::0,0,0,0.0IP::DEV::LVL::ErrorMASK::0100 *Router Config * voice translation-rule 1 rule 1 /415888\(1...\)/ /\1/ voice translation-profile INBOUND translate called 1* *voice-port 0/2/1:23 translation-profile incoming INBOUND* *dial-peer voice 3000 voip destination-pattern 1...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.11 dtmf-relay h245-alphanumeric no vad dial-peer voice 3005 voip preference 1 destination-pattern 1...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric no vad BR1#sh run int fa0/0.240 Building configuration... Current configuration : 204 bytes ! interface FastEthernet0/0.240 encapsulation dot1Q 240 ip address 10.10.201.1 255.255.255.0 ip helper-address 10.10.200.3 h323-gateway voip interface h323-gateway voip bind srcaddr 10.10.201.1 Weird thing is i can use csim start to call phones and they ring but inbound PSTN calls gets the error msg above any thots? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] cBarge on CME ????
I dont know which lab. i only tried out Mike Brooks's question on my lab You can check through all the labs to confirm Regards On Sun, Mar 28, 2010 at 5:12 PM, CCIETalk.com cciet...@gmail.com wrote: Trying to remember which lab was this? On 3/28/10, Omotayo adefilabi...@gmail.com wrote: its ok now the conference resources was not registered before now thanks On Sun, Mar 28, 2010 at 7:08 AM, Omotayo adefilabi...@gmail.com wrote: Hello , is the feature support of CIPC I configured it but when i pressed Cbarge while in In Use Remote, i get the message Failed to setup barge Rgd On Wed, Mar 24, 2010 at 4:13 AM, Mike Brooks 2xcci...@gmail.com wrote: Yes. Thanks Jason Mark. I was missing the conference hardware command under telephony-service. I appreciate your help on this. Thx, Mike Brooks CCIE#16027 (RS) On Tue, Mar 23, 2010 at 11:10 PM, Jason Granat j...@slash128.com wrote: Also, check to see if normal ad-hoc hardware conf works first before tshooting cbarge. Sent while mobile. On Mar 23, 2010, at 20:06, Mike Brooks 2xcci...@gmail.com wrote: Yes I have those still same results...? ! ephone-dn 5 octo-line number 3555 no-reg conference ad-hoc no huntstop ! ! ephone-dn 6 octo-line number 3666 no-reg conference ad-hoc preference 1 no huntstop ! ! ephone-dn 7 octo-line number 3777 no-reg conference ad-hoc preference 2 no huntstop ! ! ephone-dn 8 octo-line number 3888 no-reg conference ad-hoc preference 3 no huntstop ! On Tue, Mar 23, 2010 at 11:00 PM, Mark Nigh mn...@netelligent.com mn...@netelligent.com wrote: Yes. That's right. Ephon-dn x octel Number Conference ad-hoc Mark Nigh Systems Engineer mn...@netelligent.commn...@netelligent.com (p) 314.392.6926 -Original Message- From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat Sent: Tuesday, March 23, 2010 9:55 PM To: Mike Brooks Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] cBarge on CME Pretty sure you need some ad-hoc conf dn's Sent while mobile. On Mar 23, 2010, at 19:49, Mike Brooks 2xcci...@gmail.com 2xcci...@gmail.com wrote: On CME I am having issues with cBarge. I have a shared DN 3010 on line 2 of both ephone 1 and 2. When a call comes in on 3010 and is picked up by either phone I can see the display of remote in use. When I click on the cBarge key on the phone nothing happens. It is the same regardless of which phone is trying to barge in. I must be missing something. Any suggestions ? ! ephone-template 1 softkeys remote-in-use CBarge Newcall ! ephone-dn 10 octo-line number 3010 no-reg primary ! ephone 1 privacy off mac-address 001B.5495.1AB9 ephone-template 1 type 7961GE button 1:1 2:10 ! ephone 2 privacy off mac-address 000C.85B9.8739 ephone-template 1 type 7960 button 1:2 2:10 ! ! BR2-RTR#sho dspfarm profile 1 Dspfarm Profile Configuration Profile ID = 1, Service = CONFERENCING, Resource ID = 1 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 2 Number of Resource Available : 2 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required Codec : g729ar8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729abr8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729r8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729br8, Maximum Packetization Period : 60 , Transcoder: Not Required BR2-RTR# Thx, Mike ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/www.ipexpert.com http://slash128.com/http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/www.ipexpert.com This transmission and any attached files are privileged, confidential or otherwise the exclusive property of the intended recipient or Netelligent Corporation. If you are not the intended recipient, any disclosure, copying, distribution or use of any of the information contained in or attached to this transmission is strictly prohibited. If you have
Re: [OSL | CCIE_Voice] Discrepancy between Mark Snow's VoD and Lab Results?
they advised you call from the pstn On Sun, Mar 28, 2010 at 7:56 PM, James Key j...@jackhenry.com wrote: I think you are running into the same PL mutlicast not working over VPN. It is been quite awhile since I have used the PL racks for voice, but remember they had a reccomended way for testing mutlicast. Just cant remember. James -- *From:* Matthew Berry [ciscovoiceg...@gmail.com] *Sent:* Sunday, March 28, 2010 1:49 PM *To:* James Key *Cc:* OSL *Subject:* Re: [OSL | CCIE_Voice] Discrepancy between Mark Snow's VoD and Lab Results? James - The command is not in my config. I attached it for reference. When I run a *debug ephone moh* I can see the MoH routes being used. However, if I run a *show ccm-manager music-on-hold* while the PSTN caller is on hold, I do not see the session count increment from 0 to 1. This seems off. Matthew Berry *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* *Gmail:* ciscovoiceguru *Skype:* ciscovoiceguru *Twitter:* ciscovoiceguru *1st Lab Attempt: *Aug 16, 2010 On 3/28/2010 1:38 PM, James Key wrote: Do you currently have multicast MOH streaming from flash? If so, make sure the no mgcp timer receive-rtcp is NOT in your config. If it is, remove (don't forget no mgcp/mgcp ;-) ). Place a call in from pstn and place on hold. Your call should disconnect after the given time. James Key -- *From:* Matthew Berry [ciscovoiceg...@gmail.com] *Sent:* Sunday, March 28, 2010 12:45 PM *To:* James Key *Cc:* OSL *Subject:* Re: [OSL | CCIE_Voice] Discrepancy between Mark Snow's VoD and Lab Results? James, I see what you're saying here. Could you recommend a good test scenario to see this 30 sec disconnect? Right now, my phones are looking at 239.1.1.1 for their MOH, the CUCM Sub is setup to send MMOH out over that IP, but I am blocking multicast from traversing the WAN. I also have SRST setup with the proper MOH commands. I thought that was enough to test this scenario. Is there something I am missing? Matthew Berry *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* *Gmail:* ciscovoiceguru *Skype:* ciscovoiceguru *Twitter:* ciscovoiceguru *1st Lab Attempt: *Aug 16, 2010 On 3/28/2010 12:17 PM, James Key wrote: Mark is correct here. The command is needed when sourcing MOH from flash on am mgcp gateway. As soon as call is placed on hold, mgcp traffic stops traversing the wan and the call will drop at 30 seconds. When multicasting across the WAN as you are doing, no issues. Was able to reproduce this in the lab during my studies. James Key -- *From:* ccie_voice-boun...@onlinestudylist.com [ ccie_voice-boun...@onlinestudylist.com] On Behalf Of Matthew Berry [ ciscovoiceg...@gmail.com] *Sent:* Sunday, March 28, 2010 11:13 AM *To:* OSL *Subject:* [OSL | CCIE_Voice] Discrepancy between Mark Snow's VoD and Lab Results? The following is an excerpt from Mark Snow's V3 VoD released late last year. · *If on an MGCP gateway: “no mgcp timer receive-rtcp” *– If you have an MGCP gateway at a remote site and the requirement is to have a no-WAN MoH solution, you must enter this command. oYou will be sourcing the MoH locally on a SCCP “server” (SRST gateway) and no MGCP traffic will be traversing the WAN. oOtherwise, you will be on hold for exactly 30 seconds. When testing, stay on hold for at least 1 minute. I just went through Vol 1 Lab 7 Question 7.2, where you setup MMOH over the WAN to BR1. I did some testing and verified that this command was not needed to maintain a connection between the PSTN phone and BR1 Phone 2. Has anyone else seen this? -- Matthew Berry *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* *Gmail:* ciscovoiceguru *Skype:* ciscovoiceguru *Twitter:* ciscovoiceguru *1st Lab Attempt: *Aug 16, 2010 NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. NOTICE: This electronic mail message and any files transmitted with it
Re: [OSL | CCIE_Voice] cBarge on CME ????
Hello , is the feature support of CIPC I configured it but when i pressed Cbarge while in In Use Remote, i get the message Failed to setup barge Rgd On Wed, Mar 24, 2010 at 4:13 AM, Mike Brooks 2xcci...@gmail.com wrote: Yes. Thanks Jason Mark. I was missing the conference hardware command under telephony-service. I appreciate your help on this. Thx, Mike Brooks CCIE#16027 (RS) On Tue, Mar 23, 2010 at 11:10 PM, Jason Granat j...@slash128.com wrote: Also, check to see if normal ad-hoc hardware conf works first before tshooting cbarge. Sent while mobile. On Mar 23, 2010, at 20:06, Mike Brooks 2xcci...@gmail.com wrote: Yes I have those still same results...? ! ephone-dn 5 octo-line number 3555 no-reg conference ad-hoc no huntstop ! ! ephone-dn 6 octo-line number 3666 no-reg conference ad-hoc preference 1 no huntstop ! ! ephone-dn 7 octo-line number 3777 no-reg conference ad-hoc preference 2 no huntstop ! ! ephone-dn 8 octo-line number 3888 no-reg conference ad-hoc preference 3 no huntstop ! On Tue, Mar 23, 2010 at 11:00 PM, Mark Nigh mn...@netelligent.com mn...@netelligent.com wrote: Yes. That's right. Ephon-dn x octel Number Conference ad-hoc Mark Nigh Systems Engineer mn...@netelligent.commn...@netelligent.com (p) 314.392.6926 -Original Message- From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat Sent: Tuesday, March 23, 2010 9:55 PM To: Mike Brooks Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] cBarge on CME Pretty sure you need some ad-hoc conf dn's Sent while mobile. On Mar 23, 2010, at 19:49, Mike Brooks 2xcci...@gmail.com 2xcci...@gmail.com wrote: On CME I am having issues with cBarge. I have a shared DN 3010 on line 2 of both ephone 1 and 2. When a call comes in on 3010 and is picked up by either phone I can see the display of remote in use. When I click on the cBarge key on the phone nothing happens. It is the same regardless of which phone is trying to barge in. I must be missing something. Any suggestions ? ! ephone-template 1 softkeys remote-in-use CBarge Newcall ! ephone-dn 10 octo-line number 3010 no-reg primary ! ephone 1 privacy off mac-address 001B.5495.1AB9 ephone-template 1 type 7961GE button 1:1 2:10 ! ephone 2 privacy off mac-address 000C.85B9.8739 ephone-template 1 type 7960 button 1:2 2:10 ! ! BR2-RTR#sho dspfarm profile 1 Dspfarm Profile Configuration Profile ID = 1, Service = CONFERENCING, Resource ID = 1 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 2 Number of Resource Available : 2 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required Codec : g729ar8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729abr8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729r8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729br8, Maximum Packetization Period : 60 , Transcoder: Not Required BR2-RTR# Thx, Mike ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/www.ipexpert.com http://slash128.com/http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/www.ipexpert.com This transmission and any attached files are privileged, confidential or otherwise the exclusive property of the intended recipient or Netelligent Corporation. If you are not the intended recipient, any disclosure, copying, distribution or use of any of the information contained in or attached to this transmission is strictly prohibited. If you have received this transmission in error, please contact us immediately by responding to this message or by telephone (314-392-6900) and promptly destroy the original transmission and its attachments. -- http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] cBarge on CME ????
its ok now the conference resources was not registered before now thanks On Sun, Mar 28, 2010 at 7:08 AM, Omotayo adefilabi...@gmail.com wrote: Hello , is the feature support of CIPC I configured it but when i pressed Cbarge while in In Use Remote, i get the message Failed to setup barge Rgd On Wed, Mar 24, 2010 at 4:13 AM, Mike Brooks 2xcci...@gmail.com wrote: Yes. Thanks Jason Mark. I was missing the conference hardware command under telephony-service. I appreciate your help on this. Thx, Mike Brooks CCIE#16027 (RS) On Tue, Mar 23, 2010 at 11:10 PM, Jason Granat j...@slash128.com wrote: Also, check to see if normal ad-hoc hardware conf works first before tshooting cbarge. Sent while mobile. On Mar 23, 2010, at 20:06, Mike Brooks 2xcci...@gmail.com wrote: Yes I have those still same results...? ! ephone-dn 5 octo-line number 3555 no-reg conference ad-hoc no huntstop ! ! ephone-dn 6 octo-line number 3666 no-reg conference ad-hoc preference 1 no huntstop ! ! ephone-dn 7 octo-line number 3777 no-reg conference ad-hoc preference 2 no huntstop ! ! ephone-dn 8 octo-line number 3888 no-reg conference ad-hoc preference 3 no huntstop ! On Tue, Mar 23, 2010 at 11:00 PM, Mark Nigh mn...@netelligent.com mn...@netelligent.com wrote: Yes. That's right. Ephon-dn x octel Number Conference ad-hoc Mark Nigh Systems Engineer mn...@netelligent.commn...@netelligent.com (p) 314.392.6926 -Original Message- From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat Sent: Tuesday, March 23, 2010 9:55 PM To: Mike Brooks Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] cBarge on CME Pretty sure you need some ad-hoc conf dn's Sent while mobile. On Mar 23, 2010, at 19:49, Mike Brooks 2xcci...@gmail.com 2xcci...@gmail.com wrote: On CME I am having issues with cBarge. I have a shared DN 3010 on line 2 of both ephone 1 and 2. When a call comes in on 3010 and is picked up by either phone I can see the display of remote in use. When I click on the cBarge key on the phone nothing happens. It is the same regardless of which phone is trying to barge in. I must be missing something. Any suggestions ? ! ephone-template 1 softkeys remote-in-use CBarge Newcall ! ephone-dn 10 octo-line number 3010 no-reg primary ! ephone 1 privacy off mac-address 001B.5495.1AB9 ephone-template 1 type 7961GE button 1:1 2:10 ! ephone 2 privacy off mac-address 000C.85B9.8739 ephone-template 1 type 7960 button 1:2 2:10 ! ! BR2-RTR#sho dspfarm profile 1 Dspfarm Profile Configuration Profile ID = 1, Service = CONFERENCING, Resource ID = 1 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 2 Number of Resource Available : 2 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required Codec : g729ar8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729abr8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729r8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729br8, Maximum Packetization Period : 60 , Transcoder: Not Required BR2-RTR# Thx, Mike ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/www.ipexpert.com http://slash128.com/http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/www.ipexpert.com This transmission and any attached files are privileged, confidential or otherwise the exclusive property of the intended recipient or Netelligent Corporation. If you are not the intended recipient, any disclosure, copying, distribution or use of any of the information contained in or attached to this transmission is strictly prohibited. If you have received this transmission in error, please contact us immediately by responding to this message or by telephone (314-392-6900) and promptly destroy the original transmission and its attachments. -- http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding
Re: [OSL | CCIE_Voice] SIP Integration of UCM and UC
yes i did On Thu, Mar 25, 2010 at 4:22 AM, Otto Sanchez o...@ipexpert.com wrote: Hello, Did you also checked that: 1.- Sip trunk security profile has Accept Unsolicited Notification checked 2.- Some ports in UC are enabled to Send MWI Requests Thanks, On Mon, Mar 22, 2010 at 11:40 PM, Omotayo adefilabi...@gmail.com wrote: Hello Otto, I checked the Redirecting Diversion Header Delivery - Inbound and Redirecting Diversion Header Delivery - outbound Voicemail works now but MWI is not working what do i need to do to fix it thanks On Mon, Mar 22, 2010 at 10:56 AM, Omotayo adefilabi...@gmail.com wrote: Hello, That should be on the sip trunk right? I am not sure i checked that. i will confirm today and give you update Regards On Mon, Mar 22, 2010 at 2:07 AM, Otto Sanchez o...@ipexpert.comwrote: I meant for the *Out*bound direction, i.e., from ucm to uc, On Sun, Mar 21, 2010 at 5:51 PM, Otto Sanchez o...@ipexpert.comwrote: Hi, Did you take a look at this document? http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.htmlAlso, make sure the Redirecting Diversion Header Delivery - Inbound is checked, hth, On Fri, Mar 19, 2010 at 1:35 PM, Omotayo adefilabi...@gmail.comwrote: i have been able to get this work. i have checked all doc but no solution I still need help on this thanks On Wed, Mar 17, 2010 at 3:29 PM, Omotayo adefilabi...@gmail.comwrote: Hello, Any ideas? On Wed, Mar 17, 2010 at 9:54 AM, Omotayo adefilabi...@gmail.comwrote: Hello All, On Lab 7, after integrating the UCM with the UC using SIP. Pressing the subscriber button, i get the personal greeting message But, when pstn or a local call dials hq phone 2 or br1 phone 2, i hear Hello Cisco unity connection messaging system from a text tone phone. Any one with an idea why this i s happening NB: I deleted all the preconfigured voicemail port, huntlist, hunt group and hunt pilot on the UCM as the gude does not indicate that it is needed for the integration to wor Thanks for the anticipated response Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Integration of UCM and UC
Hello, That should be on the sip trunk right? I am not sure i checked that. i will confirm today and give you update Regards On Mon, Mar 22, 2010 at 2:07 AM, Otto Sanchez o...@ipexpert.com wrote: I meant for the *Out*bound direction, i.e., from ucm to uc, On Sun, Mar 21, 2010 at 5:51 PM, Otto Sanchez o...@ipexpert.com wrote: Hi, Did you take a look at this document? http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.htmlAlso, make sure the Redirecting Diversion Header Delivery - Inbound is checked, hth, On Fri, Mar 19, 2010 at 1:35 PM, Omotayo adefilabi...@gmail.comwrote: i have been able to get this work. i have checked all doc but no solution I still need help on this thanks On Wed, Mar 17, 2010 at 3:29 PM, Omotayo adefilabi...@gmail.comwrote: Hello, Any ideas? On Wed, Mar 17, 2010 at 9:54 AM, Omotayo adefilabi...@gmail.comwrote: Hello All, On Lab 7, after integrating the UCM with the UC using SIP. Pressing the subscriber button, i get the personal greeting message But, when pstn or a local call dials hq phone 2 or br1 phone 2, i hear Hello Cisco unity connection messaging system from a text tone phone. Any one with an idea why this i s happening NB: I deleted all the preconfigured voicemail port, huntlist, hunt group and hunt pilot on the UCM as the gude does not indicate that it is needed for the integration to wor Thanks for the anticipated response Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Integration of UCM and UC
Hello Otto, I checked the Redirecting Diversion Header Delivery - Inbound and Redirecting Diversion Header Delivery - outbound Voicemail works now but MWI is not working what do i need to do to fix it thanks On Mon, Mar 22, 2010 at 10:56 AM, Omotayo adefilabi...@gmail.com wrote: Hello, That should be on the sip trunk right? I am not sure i checked that. i will confirm today and give you update Regards On Mon, Mar 22, 2010 at 2:07 AM, Otto Sanchez o...@ipexpert.com wrote: I meant for the *Out*bound direction, i.e., from ucm to uc, On Sun, Mar 21, 2010 at 5:51 PM, Otto Sanchez o...@ipexpert.com wrote: Hi, Did you take a look at this document? http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.htmlAlso, make sure the Redirecting Diversion Header Delivery - Inbound is checked, hth, On Fri, Mar 19, 2010 at 1:35 PM, Omotayo adefilabi...@gmail.comwrote: i have been able to get this work. i have checked all doc but no solution I still need help on this thanks On Wed, Mar 17, 2010 at 3:29 PM, Omotayo adefilabi...@gmail.comwrote: Hello, Any ideas? On Wed, Mar 17, 2010 at 9:54 AM, Omotayo adefilabi...@gmail.comwrote: Hello All, On Lab 7, after integrating the UCM with the UC using SIP. Pressing the subscriber button, i get the personal greeting message But, when pstn or a local call dials hq phone 2 or br1 phone 2, i hear Hello Cisco unity connection messaging system from a text tone phone. Any one with an idea why this i s happening NB: I deleted all the preconfigured voicemail port, huntlist, hunt group and hunt pilot on the UCM as the gude does not indicate that it is needed for the integration to wor Thanks for the anticipated response Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Integration of UCM and UC
i have been able to get this work. i have checked all doc but no solution I still need help on this thanks On Wed, Mar 17, 2010 at 3:29 PM, Omotayo adefilabi...@gmail.com wrote: Hello, Any ideas? On Wed, Mar 17, 2010 at 9:54 AM, Omotayo adefilabi...@gmail.com wrote: Hello All, On Lab 7, after integrating the UCM with the UC using SIP. Pressing the subscriber button, i get the personal greeting message But, when pstn or a local call dials hq phone 2 or br1 phone 2, i hear Hello Cisco unity connection messaging system from a text tone phone. Any one with an idea why this i s happening NB: I deleted all the preconfigured voicemail port, huntlist, hunt group and hunt pilot on the UCM as the gude does not indicate that it is needed for the integration to wor Thanks for the anticipated response Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling
No I only change the Single button barge to CBarge On Wed, Mar 17, 2010 at 9:49 PM, Radhesh Naik radheshn...@gmail.com wrote: Is you built in bridge in phone configuration set to on by any chance ? even if its set to default try changing it to off. -Radhesh *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* Wednesday, March 17, 2010 8:21 PM *To:* Peter Slow *Cc:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling yes i applied the mgl to the br2 device pool On Wed, Mar 17, 2010 at 9:20 PM, Peter Slow peter.s...@gmail.com wrote: do you mean device pool or MRGL? No CFB means CUCM couldnt select a CFB from the MRGL it was using for selection. -Peter On Wed, Mar 17, 2010 at 10:46 AM, Omotayo adefilabi...@gmail.com wrote: Hello, i meant to say i put all the software conference brige in a device pool that is not assigned to any user thanks On Wed, Mar 17, 2010 at 9:09 AM, Omotayo adefilabi...@gmail.com wrote: Hello All, Worked on Lab 7 question 3 - DISA daling. i had two issues with this section while working on it; Q3.3. i configured conference resources on the br2 gateway and applied to the BR2 Device pool. While the phone is In Remote Use . Also applied CBarge on the BR2 phones. On pressing the red button, i get a reorder tone on both Br2 phone and the Hq phone After this i put all the Hardware conference brige in a device pool that is not assigned to any user, this time, when i press the red button, i get No Conference Bridge Anyone with an idea what the issue is because i have the software conference bridge registered on the UCM Also i want to know why i can not apply 07976852817 as the remote destination profile with partial match set at the servie parameter. i did and the br2 phone did not blink when calling HQ or BR1 phone. it was when i added the +447976852817 that it worked Thanks for the anticipated response ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] about globalization and the lab's PSTN
I always wonder if the guyz that wrote the PG looks through all this concens. not just for this question but for all They should help us with some of this issues as some of us trust that all that is on the PG is what is needed, though i check config guides atimes On Thu, Mar 18, 2010 at 3:17 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: That's correct. When the call is delivered to the HQ gateway it is seen as national from the perspective of the terminating ISDN. From the standpoint of HQ, the 212 is national type. If it was a local PRI from a local LEC, I might expect a subscriber type, but itd be most common to receive a national type. - Sent from my Blackberry -- *From*: Jean M. Thewissen m...@mnet.com.mx *To*: Berry, Matthew J.; 'ccie_voice@onlinestudylist.com' ccie_voice@onlinestudylist.com *Sent*: Wed Mar 17 20:55:13 2010 *Subject*: RE: [OSL | CCIE_Voice] about globalization and the lab's PSTN I am using proctorlabs… What really confuses me is that when the call leaves the B2 GW, it is correctly tagged as international. Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'b2 phone 3' Calling Party Number i = 0x0081, '+3432143003' Plan:Unknown, Type:Unknown *Called Party Number i = 0x91, '0012123945001'* *Plan:ISDN, Type:International* But when PSTN sends it to HQ GW, it is tagged as national: Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Display i = 'b2 phone 3' Calling Party Number i = 0x0081, '+3432143003' Plan:Unknown, Type:Unknown *Called Party Number i = 0xA1, '2123945001'* *Plan:ISDN, Type:National* I really don’t see how I could alter how the PSTN tags the call… but maybe I am just not seeing the whole picture. *From:* Berry, Matthew J. [mailto:mjbe...@krollontrack.com] *Sent:* miércoles, 17 de marzo de 2010 07:50 p.m. *To:* Jean M. Thewissen; 'ccie_voice@onlinestudylist.com' *Subject:* Re: [OSL | CCIE_Voice] about globalization and the lab's PSTN Jean, Are you using Proctor Labs or your own lab? The PSTN router should take care of the calling number type. You should also make sure you don't have any translation patterns on the BR2 gateway that would modify the type. Also check your H323 gateway to ensure the same thing. - Sent from my Blackberry -- *From*: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com *To*: 'ccie_voice@onlinestudylist.com' ccie_voice@onlinestudylist.com *Sent*: Wed Mar 17 19:21:33 2010 *Subject*: [OSL | CCIE_Voice] about globalization and the lab's PSTN Hello All. I am experiencing the following behavior: I place a call out of the Brach 2 site, internationally into the HQ site, the PSTN sends the call into HQ as “national”. If I place the call from the PSTN phone international line (India or Spain) into the HQ site, the call comes in correctly labeled as “international”. Is this the expected behavior due to the simulation of the PSTN in the lab? Or I am not setting something that I should when routing the calls out the Branch2 site? Any advice is greatly appreciated. Regards! MT ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] [OSL | CCIE_RS] [OSL | CCIE_Security] Great New Offers from IPexpert
Hello, Has this 4 point OEQ been effected On Thu, Mar 18, 2010 at 12:23 PM, Mike Thompson mthompson...@gmail.comwrote: ??? When did that change take place?and why? Sent from my phone, apologies for any typos. On Mar 18, 2010, at 4:11 AM, Paul Kruger pauld.kru...@gmail.com wrote: Hi Mike, While your logic seems air-tight, I just want to let you guys know that you shouldn't count on the OEQ's giving you 21 points anymore. They've changed it to 4 points only. At least for the Voice Lab. I can't vouch for the other tracks. But! It is still mandatory pass this section to pass the lab. Even if it is only 4 points. If you fail this and get full 96 points from the lab, you still fail. Keep that in mind. It's an important bit to know, as it makes the test harder. I had my second attempt at the end of Jan, and this was implemented already. Third attempt: Aug/Sep. On Thu, Mar 18, 2010 at 2:14 AM, Mike Thompson mthompson...@gmail.comwrote: Everyone keeps talking about the ‘free OEQ’, but they’re forgetting a key thing. The OEQ is 30 minutes. There are 4 questions that are, if you’re actually prepared for the exam, elementary questions. Each of these questions take a few minutes to answer on average. I mean there isn’t much calculation to ‘what is the TCP port used to control an IP Phone’ (not an actual question I received). It’s not like they want you to calculate the airspeed of an unlaiden swallow. In all reality, the 4 OEQ questions take less than 15 minutes and give you a 21 point head start on your exam (and yes, it’s not really a head start, but you get the gist). More importantly, you get an extra 15 minutes to get that critical 69 or more points to get your number. Would you rather spend 480 minutes getting 100 points (.2083 points per minute) Or 465 minutes getting 89 points (.1914 points per minute). Just my 2 cents folks *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Tyson Scott *Sent:* Wednesday, March 17, 2010 1:46 AM *To:* 'Nadeem Rafi'; 'Brandon Carroll' *Cc:* 'CCIE_RS OnlineStudyList'; 'ccie OSL'; ccie...@onlinestudylist.com; ccie_secur...@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] [OSL | CCIE_RS] [OSL | CCIE_Security] Great New Offers from IPexpert But wait haha. Thanks Marko. Regards, Tyson Scott - CCIE #13513 RS, Security, and SP Technical Instructor - IPexpert, Inc. Mailto: tsc...@ipexpert.com Telephone: +1.810.326.1444, ext. 208 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com *From:* ccie_rs-boun...@onlinestudylist.com [mailto: ccie_rs-boun...@onlinestudylist.com] *On Behalf Of *Nadeem Rafi *Sent:* Wednesday, March 17, 2010 12:53 AM *To:* Brandon Carroll *Cc:* CCIE_RS OnlineStudyList; ccie OSL; ccie_secur...@onlinestudylist.com; ccie...@onlinestudylist.com *Subject:* Re: [OSL | CCIE_RS] [OSL | CCIE_Security] Great New Offers from IPexpert a good answer to purchase your ccie kind stuff. On Wed, Mar 17, 2010 at 7:47 AM, Brandon Carroll bcarr...@ipexpert.com wrote: Really Marko, when you step back and look at it this just reenforces what we all know: Time on the racks is more valuable than anything when you are preparing for the CCIE. Regards, Brandon Carroll - CCIE #23837 Senior Technical Instructor - IPexpert Mailto: bcarr...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com On Tue, Mar 16, 2010 at 9:45 PM, Marko Milivojevic mar...@ipexpert.com wrote: On Wed, Mar 17, 2010 at 04:39, Brandon Carroll bcarr...@ipexpert.com wrote: Do you know what 400 hours of rack time cost me when I was a student? Not only from an instructor point of view, but also from a former students point of viewWOW. Unbelievable! Yeah. It's a $1400 right there, only in rack time. We're practically giving away this deal. -- Marko Milivojevic - CCIE #18427 Senior Technical Instructor - IPexpert YES! We include 400 hours of REAL rack time with our Blended Learning
Re: [OSL | CCIE_Voice] [OSL | CCIE_RS] [OSL | CCIE_Security] Great New Offers from IPexpert
Hello, is this information available online On Thu, Mar 18, 2010 at 2:20 PM, Marko Milivojevic mar...@ipexpert.comwrote: On Thu, Mar 18, 2010 at 11:23, Mike Thompson mthompson...@gmail.com wrote: ??? When did that change take place?and why? It changed starting January 4th, if I recall. The way one of the proctors explained it to me, they thought the OEQs with 21 points carried way too many points, compared to the rest of the lab. They simply changed the rule to say you need to pass both sections, instead of treating OEQs as the part of the overall score. It still doesn't matter - you need 80% of the points in all sections. Figuring out what makes 80% is a little bit more dfficult now that total score isn't 100. -- Marko Milivojevic - CCIE #18427 Senior Technical Instructor - IPexpert YES! We include 400 hours of REAL rack time with our Blended Learning Solution! Mailto: mar...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Web: http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] [OSL | CCIE_RS] [OSL | CCIE_Security] Great New Offers from IPexpert
Hello, Just think it wont hurt to have an idea of what is going on Thanks for the info Regards On Thu, Mar 18, 2010 at 2:49 PM, Marko Milivojevic mar...@ipexpert.comwrote: On Thu, Mar 18, 2010 at 13:25, Omotayo adefilabi...@gmail.com wrote: Hello, is this information available online Not that I know of. Why does it matter, though? You still need to pass both sections, nothing changes, really. -- Marko Milivojevic - CCIE #18427 Senior Technical Instructor - IPexpert YES! We include 400 hours of REAL rack time with our Blended Learning Solution! Mailto: mar...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Web: http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling
Hello, i set the in built bridge to On on the hq phone 2( phone that is barged into). i get the Entance party tone and a message you are not a valid conference partipant i tried again i got the entrance party tone and message No Conference bridge On Thu, Mar 18, 2010 at 11:02 AM, Omotayo adefilabi...@gmail.com wrote: No I only change the Single button barge to CBarge On Wed, Mar 17, 2010 at 9:49 PM, Radhesh Naik radheshn...@gmail.comwrote: Is you built in bridge in phone configuration set to on by any chance ? even if its set to default try changing it to off. -Radhesh *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* Wednesday, March 17, 2010 8:21 PM *To:* Peter Slow *Cc:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling yes i applied the mgl to the br2 device pool On Wed, Mar 17, 2010 at 9:20 PM, Peter Slow peter.s...@gmail.com wrote: do you mean device pool or MRGL? No CFB means CUCM couldnt select a CFB from the MRGL it was using for selection. -Peter On Wed, Mar 17, 2010 at 10:46 AM, Omotayo adefilabi...@gmail.com wrote: Hello, i meant to say i put all the software conference brige in a device pool that is not assigned to any user thanks On Wed, Mar 17, 2010 at 9:09 AM, Omotayo adefilabi...@gmail.com wrote: Hello All, Worked on Lab 7 question 3 - DISA daling. i had two issues with this section while working on it; Q3.3. i configured conference resources on the br2 gateway and applied to the BR2 Device pool. While the phone is In Remote Use . Also applied CBarge on the BR2 phones. On pressing the red button, i get a reorder tone on both Br2 phone and the Hq phone After this i put all the Hardware conference brige in a device pool that is not assigned to any user, this time, when i press the red button, i get No Conference Bridge Anyone with an idea what the issue is because i have the software conference bridge registered on the UCM Also i want to know why i can not apply 07976852817 as the remote destination profile with partial match set at the servie parameter. i did and the br2 phone did not blink when calling HQ or BR1 phone. it was when i added the +447976852817 that it worked Thanks for the anticipated response ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling
Hello, Am sure the conference resource on the br2 router is reachable because i tested the Join Across Line feature on Br2 phone and the conference resource was invoked I do not know why the Barge is not working BR2-RTR#sh sccp connections sess_idconn_idstype mode codec ripaddr rport sport 33555434 33554486 conf sendrecv g729192.168.3.1225860 18686 33555434 33554484 conf sendrecv g729192.168.3.1420442 18788 33555434 33554482 conf sendrecv g711u 192.168.3.1618172 17154 On Fri, Mar 19, 2010 at 12:34 AM, Omotayo adefilabi...@gmail.com wrote: Hello, i set the in built bridge to On on the hq phone 2( phone that is barged into). i get the Entance party tone and a message you are not a valid conference partipant i tried again i got the entrance party tone and message No Conference bridge On Thu, Mar 18, 2010 at 11:02 AM, Omotayo adefilabi...@gmail.comwrote: No I only change the Single button barge to CBarge On Wed, Mar 17, 2010 at 9:49 PM, Radhesh Naik radheshn...@gmail.comwrote: Is you built in bridge in phone configuration set to on by any chance ? even if its set to default try changing it to off. -Radhesh *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* Wednesday, March 17, 2010 8:21 PM *To:* Peter Slow *Cc:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling yes i applied the mgl to the br2 device pool On Wed, Mar 17, 2010 at 9:20 PM, Peter Slow peter.s...@gmail.com wrote: do you mean device pool or MRGL? No CFB means CUCM couldnt select a CFB from the MRGL it was using for selection. -Peter On Wed, Mar 17, 2010 at 10:46 AM, Omotayo adefilabi...@gmail.com wrote: Hello, i meant to say i put all the software conference brige in a device pool that is not assigned to any user thanks On Wed, Mar 17, 2010 at 9:09 AM, Omotayo adefilabi...@gmail.com wrote: Hello All, Worked on Lab 7 question 3 - DISA daling. i had two issues with this section while working on it; Q3.3. i configured conference resources on the br2 gateway and applied to the BR2 Device pool. While the phone is In Remote Use . Also applied CBarge on the BR2 phones. On pressing the red button, i get a reorder tone on both Br2 phone and the Hq phone After this i put all the Hardware conference brige in a device pool that is not assigned to any user, this time, when i press the red button, i get No Conference Bridge Anyone with an idea what the issue is because i have the software conference bridge registered on the UCM Also i want to know why i can not apply 07976852817 as the remote destination profile with partial match set at the servie parameter. i did and the br2 phone did not blink when calling HQ or BR1 phone. it was when i added the +447976852817 that it worked Thanks for the anticipated response ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling
Hello All, Worked on Lab 7 question 3 - DISA daling. i had two issues with this section while working on it; Q3.3. i configured conference resources on the br2 gateway and applied to the BR2 Device pool. While the phone is In Remote Use . Also applied CBarge on the BR2 phones. On pressing the red button, i get a reorder tone on both Br2 phone and the Hq phone After this i put all the Hardware conference brige in a device pool that is not assigned to any user, this time, when i press the red button, i get No Conference Bridge Anyone with an idea what the issue is because i have the software conference bridge registered on the UCM Also i want to know why i can not apply 07976852817 as the remote destination profile with partial match set at the servie parameter. i did and the br2 phone did not blink when calling HQ or BR1 phone. it was when i added the +447976852817 that it worked Thanks for the anticipated response ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] SIP Integration of UCM and UC
Hello All, On Lab 7, after integrating the UCM with the UC using SIP. Pressing the subscriber button, i get the personal greeting message But, when pstn or a local call dials hq phone 2 or br1 phone 2, i hear Hello Cisco unity connection messaging system from a text tone phone. Any one with an idea why this i s happening NB: I deleted all the preconfigured voicemail port, huntlist, hunt group and hunt pilot on the UCM as the gude does not indicate that it is needed for the integration to wor Thanks for the anticipated response Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling
Any one with a fix on this? On Wed, Mar 17, 2010 at 9:09 AM, Omotayo adefilabi...@gmail.com wrote: Hello All, Worked on Lab 7 question 3 - DISA daling. i had two issues with this section while working on it; Q3.3. i configured conference resources on the br2 gateway and applied to the BR2 Device pool. While the phone is In Remote Use . Also applied CBarge on the BR2 phones. On pressing the red button, i get a reorder tone on both Br2 phone and the Hq phone After this i put all the Hardware conference brige in a device pool that is not assigned to any user, this time, when i press the red button, i get No Conference Bridge Anyone with an idea what the issue is because i have the software conference bridge registered on the UCM Also i want to know why i can not apply 07976852817 as the remote destination profile with partial match set at the servie parameter. i did and the br2 phone did not blink when calling HQ or BR1 phone. it was when i added the +447976852817 that it worked Thanks for the anticipated response ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Integration of UCM and UC
Hello, Any ideas? On Wed, Mar 17, 2010 at 9:54 AM, Omotayo adefilabi...@gmail.com wrote: Hello All, On Lab 7, after integrating the UCM with the UC using SIP. Pressing the subscriber button, i get the personal greeting message But, when pstn or a local call dials hq phone 2 or br1 phone 2, i hear Hello Cisco unity connection messaging system from a text tone phone. Any one with an idea why this i s happening NB: I deleted all the preconfigured voicemail port, huntlist, hunt group and hunt pilot on the UCM as the gude does not indicate that it is needed for the integration to wor Thanks for the anticipated response Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling
Hello, i meant to say i put all the software conference brige in a device pool that is not assigned to any user thanks On Wed, Mar 17, 2010 at 9:09 AM, Omotayo adefilabi...@gmail.com wrote: Hello All, Worked on Lab 7 question 3 - DISA daling. i had two issues with this section while working on it; Q3.3. i configured conference resources on the br2 gateway and applied to the BR2 Device pool. While the phone is In Remote Use . Also applied CBarge on the BR2 phones. On pressing the red button, i get a reorder tone on both Br2 phone and the Hq phone After this i put all the Hardware conference brige in a device pool that is not assigned to any user, this time, when i press the red button, i get No Conference Bridge Anyone with an idea what the issue is because i have the software conference bridge registered on the UCM Also i want to know why i can not apply 07976852817 as the remote destination profile with partial match set at the servie parameter. i did and the br2 phone did not blink when calling HQ or BR1 phone. it was when i added the +447976852817 that it worked Thanks for the anticipated response ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 7 Question 3- DISA Dialling
yes i applied the mgl to the br2 device pool On Wed, Mar 17, 2010 at 9:20 PM, Peter Slow peter.s...@gmail.com wrote: do you mean device pool or MRGL? No CFB means CUCM couldnt select a CFB from the MRGL it was using for selection. -Peter On Wed, Mar 17, 2010 at 10:46 AM, Omotayo adefilabi...@gmail.com wrote: Hello, i meant to say i put all the software conference brige in a device pool that is not assigned to any user thanks On Wed, Mar 17, 2010 at 9:09 AM, Omotayo adefilabi...@gmail.com wrote: Hello All, Worked on Lab 7 question 3 - DISA daling. i had two issues with this section while working on it; Q3.3. i configured conference resources on the br2 gateway and applied to the BR2 Device pool. While the phone is In Remote Use . Also applied CBarge on the BR2 phones. On pressing the red button, i get a reorder tone on both Br2 phone and the Hq phone After this i put all the Hardware conference brige in a device pool that is not assigned to any user, this time, when i press the red button, i get No Conference Bridge Anyone with an idea what the issue is because i have the software conference bridge registered on the UCM Also i want to know why i can not apply 07976852817 as the remote destination profile with partial match set at the servie parameter. i did and the br2 phone did not blink when calling HQ or BR1 phone. it was when i added the +447976852817 that it worked Thanks for the anticipated response ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.com wrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can’t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja -- *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* Friday, March 12, 2010 12:33 PM *To:* Vishal Preenja *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hello Otto, Yes requirement is to transport g729 over the WAN if i want to transcoder on the trunk. What do i need to do because quetion says use the hq resources The last time i applied the transcoder to the trunk, When hq phone call the sip phone on br2, i get a reorder tone When the sip phone on the br2 calls the hq phone, it disconnects on pick up and continues to ring on the sip phone Thanks On Fri, Mar 12, 2010 at 8:37 PM, Otto Sanchez o...@ipexpert.com wrote: Hi, You want to transcode at the br2 rtr as I suppose your requirement is to transport the call using g.729 over the wan right?, if that's the case, make sure the incoming dial-peer codec is set to g.729, in that case the transcoder at br2 shoud be invoked if the sip phone codec is set to g.711, On Fri, Mar 12, 2010 at 10:03 AM, Omotayo adefilabi...@gmail.com wrote: Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.comwrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can’t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UC and cme sip integration
Hello, it work ok now I was using the wrong ip address on the unity connection all the while Thanks On Fri, Mar 12, 2010 at 8:30 AM, Flemming Ortvald f...@netdesign.dk wrote: Unity connection can do both g729 and g711, you can use “voice class codec” on “voice register dn” to expand codec support for sip. Med venlig hilsen Flemming Ortvald Network System Eng. NetDesign A/S +45 4435 8346 Tænk på miljøet inden udskrivning af denne e-post og tilknyttede vedhæftninger *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* 11 March, 2010 20:58 *To:* Flemming Ortvald *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration Hello, I have to configure a transcoder on the br2 router? Unity connection support g729 only? Rgd On Thu, Mar 11, 2010 at 8:24 PM, Flemming Ortvald f...@netdesign.dk wrote: You will need a transcoder or chnage the sip endpoints to support g.711, natively it only supports g.729 Best regards Flemming Ortvald Network System Eng. NetDesign A/S +45 4435 8346 Tænk på miljøet inden udskrivning af denne e-post og tilknyttede vedhæftninger *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* 11 March, 2010 20:07 *To:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration Hello all, As anyone been able to get the SIP integration between Unity Connection and Cme to work? I followed the Proctorlabs Guide I posted this sometime lat week and revised as advised but keep getting a reorder tone( Number Unknown) when the message button is pressed Below is the relevant configuration voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 600 min 60 voice register global mode cme source-address 10.10.110.3 port 5060 max-dn 3 max-pool 6 authenticate register mwi reg-e164 voicemail 3600 tftp-path flash: create profile sync 0006855418337003 ! voice register dn 1 number 3002 call-forward b2bua busy 3600 call-forward b2bua mailbox 3002 call-forward b2bua noan 3600 timeout 12 name br2 phone 2 no-reg label br2 phone 2 mwi ! voice register dn 2 number 3003 call-forward b2bua busy 3600 call-forward b2bua mailbox 3003 call-forward b2bua noan 3600 timeout 12 name br2 phone 3 no-reg label br2 phone 3 mwi ! voice register pool 1 id mac .. type 7941 number 1 dn 1 dtmf-relay rtp-nte username 3002 password cisco ! voice register pool 2 id mac 001F.6C7E.D6FE type 7941 number 1 dn 2 dtmf-relay rtp-nte username 3003 password cisco dial-peer voice 200 voip max-conn 1 destination-pattern 3600 session protocol sipv2 session target ipv4:10.10.210.13 dtmf-relay rtp-nte codec g711ulaw ! ! telephony-service no auto-reg-ephone em logout 0:0 0:0 0:0 max-ephones 8 max-dn 8 ip source-address 10.10.202.1 port 2000 voicemail 3600 mwi relay max-conferences 8 gain -6 transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Mar 10 2010 15:22:39 ! ! ephone-dn 1 dual-line number 3001 no-reg primary label Br2 pHone 1 name Br2 Phone 1 call-forward busy 3600 call-forward noan 3600 timeout 12 ! ! sip-ua mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp unsolicited ! ! ephone 1 device-security-mode none mac-address 001E.EC15.996D type CIPC button 1:1 ! Thanks for the anticipated support ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hello Jeff, All calls worked when i configure the xcoder on the cme The question says use the hq router resources- that is where i have issues thanks On Fri, Mar 12, 2010 at 11:33 PM, Jeff Cotter jcot...@voxns.com wrote: FYI, I was only able to get this to work using transcoder on CME. Had to match the codec between UCM trunk and incoming dial-peer on CME…then xcoder would engage on CME for the SIP phone. I have a hardware limitation in my home lab so I am not able to configure a xcoder on both UCM and CME simultaneously. *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* Friday, March 12, 2010 6:33 AM *To:* Otto Sanchez *Cc:* Jeff Cotter; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] SIP Hardware Transcoder Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.com wrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can’t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UC and cme sip integration
Hello all, As anyone been able to get the SIP integration between Unity Connection and Cme to work? I followed the Proctorlabs Guide I posted this sometime lat week and revised as advised but keep getting a reorder tone( Number Unknown) when the message button is pressed Below is the relevant configuration voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 600 min 60 voice register global mode cme source-address 10.10.110.3 port 5060 max-dn 3 max-pool 6 authenticate register mwi reg-e164 voicemail 3600 tftp-path flash: create profile sync 0006855418337003 ! voice register dn 1 number 3002 call-forward b2bua busy 3600 call-forward b2bua mailbox 3002 call-forward b2bua noan 3600 timeout 12 name br2 phone 2 no-reg label br2 phone 2 mwi ! voice register dn 2 number 3003 call-forward b2bua busy 3600 call-forward b2bua mailbox 3003 call-forward b2bua noan 3600 timeout 12 name br2 phone 3 no-reg label br2 phone 3 mwi ! voice register pool 1 id mac .. type 7941 number 1 dn 1 dtmf-relay rtp-nte username 3002 password cisco ! voice register pool 2 id mac 001F.6C7E.D6FE type 7941 number 1 dn 2 dtmf-relay rtp-nte username 3003 password cisco dial-peer voice 200 voip max-conn 1 destination-pattern 3600 session protocol sipv2 session target ipv4:10.10.210.13 dtmf-relay rtp-nte codec g711ulaw ! ! telephony-service no auto-reg-ephone em logout 0:0 0:0 0:0 max-ephones 8 max-dn 8 ip source-address 10.10.202.1 port 2000 voicemail 3600 mwi relay max-conferences 8 gain -6 transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Mar 10 2010 15:22:39 ! ! ephone-dn 1 dual-line number 3001 no-reg primary label Br2 pHone 1 name Br2 Phone 1 call-forward busy 3600 call-forward noan 3600 timeout 12 ! ! sip-ua mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp unsolicited ! ! ephone 1 device-security-mode none mac-address 001E.EC15.996D type CIPC button 1:1 ! Thanks for the anticipated support ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] New VOD shippping today?
Hello, I hope it start shipping. Also i will like to confirm if those of us that bought the one recorded by Mark Snow will get this too (CCIE Voice 3.0 Video on Demand Course Accompanying Slide and Topology Books) On Wed, Mar 10, 2010 at 4:26 PM, Steve Sarrick ssarr...@drsllc.net wrote: Just curious if there are any rumors to the new VOD shipping today based on the website date of no later than March 10th. Has anyone seen/heard anything? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] calls from hq to and from cme sip phones
Hello, When i call from hq sccp phone to cme sip phone, it rings but when i pick up. it disconnects also when i call from cme sip phone to hq (sccp and sip) phone it rings on the hq phones when i pick t disconnects and contnues ringing on the sip phone i have a transcoder configured on the trunk Any one with a fix thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
Hello, the debug shows that the codec getting to the cme is g711 because the codec byte is 160 and still it disconnects On Thu, Mar 11, 2010 at 12:03 AM, Omotayo adefilabi...@gmail.com wrote: Hello, When i call from hq sccp phone to cme sip phone, it rings but when i pick up. it disconnects also when i call from cme sip phone to hq (sccp and sip) phone it rings on the hq phones when i pick t disconnects and contnues ringing on the sip phone i have a transcoder configured on the trunk Any one with a fix thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
The trunk DP has a region that speaks g729 to hq and br1 find attached the config On Thu, Mar 11, 2010 at 12:24 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Omotayo, How do you have your regions setup in CUCM? The CUCME trunk through the HQ gateway should be placed in the HQ region. Can you also send me the HQ config as an attached file. Make sure your dspfarm has a ‘no shutdown” issued. Also, make sure your transcoder is registered to CUCM under Media Resources Transcoder. Did you also make sure the transcoder is configured as an IOS Enhanced Media Termination Point? Also, make sure you ALWAYS reset the trunk in CUCM. That will oftentimes clear out weird issues. I have learned that lesson the hard way. Matthew Berry *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* Wednesday, March 10, 2010 5:20 PM *To:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones Hello, the debug shows that the codec getting to the cme is g711 because the codec byte is 160 and still it disconnects On Thu, Mar 11, 2010 at 12:03 AM, Omotayo adefilabi...@gmail.com wrote: Hello, When i call from hq sccp phone to cme sip phone, it rings but when i pick up. it disconnects also when i call from cme sip phone to hq (sccp and sip) phone it rings on the hq phones when i pick t disconnects and contnues ringing on the sip phone i have a transcoder configured on the trunk Any one with a fix thanks HQ-RTR#sh run Building configuration... Current configuration : 4129 bytes ! ! Last configuration change at 14:34:23 pst Wed Mar 10 2010 ! NVRAM config last updated at 14:32:04 pst Wed Mar 10 2010 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ-RTR ! boot-start-marker warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 clock timezone pst -8 network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! vtp domain home.com vtp mode transparent archive log config hidekeys ! ! ! ! controller T1 0/0/0 framing esf linecode ami pri-group timeslots 1-3,24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! ! ip http server no ip http secure-server ! ! ! ! ! ! ! control-plane ! ! ! voice-port 0/0/0:23 ! ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ccm-manager music-on-hold ! mgcp mgcp call-agent 10.10.210.11 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface FastEthernet0/0.20 mgcp bind media source-interface FastEthernet0/0.20 ! mgcp profile default ! sccp local FastEthernet0/0.20 sccp ccm 10.10.210.11 identifier 1 version 5.0.1 sccp ccm 10.10.210.10 identifier 2 version 5.0.1 sccp ! sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register hqtranscoder ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP ! ! ! ! ! gatekeeper zone local
Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
yes if not itwont register to the UCM Can the transcoder be configured on the br2 router?? On Thu, Mar 11, 2010 at 12:54 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Have you issued a “no shut’ on dspfarm profile 1 transcode? *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* Wednesday, March 10, 2010 5:38 PM *To:* Berry, Matthew J. *Cc:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones The trunk DP has a region that speaks g729 to hq and br1 find attached the config On Thu, Mar 11, 2010 at 12:24 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Omotayo, How do you have your regions setup in CUCM? The CUCME trunk through the HQ gateway should be placed in the HQ region. Can you also send me the HQ config as an attached file. Make sure your dspfarm has a ‘no shutdown” issued. Also, make sure your transcoder is registered to CUCM under Media Resources Transcoder. Did you also make sure the transcoder is configured as an IOS Enhanced Media Termination Point? Also, make sure you ALWAYS reset the trunk in CUCM. That will oftentimes clear out weird issues. I have learned that lesson the hard way. Matthew Berry *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* Wednesday, March 10, 2010 5:20 PM *To:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones Hello, the debug shows that the codec getting to the cme is g711 because the codec byte is 160 and still it disconnects On Thu, Mar 11, 2010 at 12:03 AM, Omotayo adefilabi...@gmail.com wrote: Hello, When i call from hq sccp phone to cme sip phone, it rings but when i pick up. it disconnects also when i call from cme sip phone to hq (sccp and sip) phone it rings on the hq phones when i pick t disconnects and contnues ringing on the sip phone i have a transcoder configured on the trunk Any one with a fix thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
Hello, When i put the trunk in the hq DP and transcoder to the br2 router, all calls works With this i did not assign the transcoder to the trunk. My concern is that we are suppose to speak g729 from hq to br2, assigning hq DP to the trunk defeat the purpose On Thu, Mar 11, 2010 at 12:56 AM, Omotayo adefilabi...@gmail.com wrote: yes if not itwont register to the UCM Can the transcoder be configured on the br2 router?? On Thu, Mar 11, 2010 at 12:54 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Have you issued a “no shut’ on dspfarm profile 1 transcode? *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* Wednesday, March 10, 2010 5:38 PM *To:* Berry, Matthew J. *Cc:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones The trunk DP has a region that speaks g729 to hq and br1 find attached the config On Thu, Mar 11, 2010 at 12:24 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Omotayo, How do you have your regions setup in CUCM? The CUCME trunk through the HQ gateway should be placed in the HQ region. Can you also send me the HQ config as an attached file. Make sure your dspfarm has a ‘no shutdown” issued. Also, make sure your transcoder is registered to CUCM under Media Resources Transcoder. Did you also make sure the transcoder is configured as an IOS Enhanced Media Termination Point? Also, make sure you ALWAYS reset the trunk in CUCM. That will oftentimes clear out weird issues. I have learned that lesson the hard way. Matthew Berry *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* Wednesday, March 10, 2010 5:20 PM *To:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones Hello, the debug shows that the codec getting to the cme is g711 because the codec byte is 160 and still it disconnects On Thu, Mar 11, 2010 at 12:03 AM, Omotayo adefilabi...@gmail.com wrote: Hello, When i call from hq sccp phone to cme sip phone, it rings but when i pick up. it disconnects also when i call from cme sip phone to hq (sccp and sip) phone it rings on the hq phones when i pick t disconnects and contnues ringing on the sip phone i have a transcoder configured on the trunk Any one with a fix thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
Hello, i sed a 7941 sip phone instead of the xlite and all the calls worked i guess we needed the xcoder because of xlite right? On Thu, Mar 11, 2010 at 1:24 AM, Omotayo adefilabi...@gmail.com wrote: Hello, When i put the trunk in the hq DP and transcoder to the br2 router, all calls works With this i did not assign the transcoder to the trunk. My concern is that we are suppose to speak g729 from hq to br2, assigning hq DP to the trunk defeat the purpose On Thu, Mar 11, 2010 at 12:56 AM, Omotayo adefilabi...@gmail.comwrote: yes if not itwont register to the UCM Can the transcoder be configured on the br2 router?? On Thu, Mar 11, 2010 at 12:54 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Have you issued a “no shut’ on dspfarm profile 1 transcode? *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* Wednesday, March 10, 2010 5:38 PM *To:* Berry, Matthew J. *Cc:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones The trunk DP has a region that speaks g729 to hq and br1 find attached the config On Thu, Mar 11, 2010 at 12:24 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Omotayo, How do you have your regions setup in CUCM? The CUCME trunk through the HQ gateway should be placed in the HQ region. Can you also send me the HQ config as an attached file. Make sure your dspfarm has a ‘no shutdown” issued. Also, make sure your transcoder is registered to CUCM under Media Resources Transcoder. Did you also make sure the transcoder is configured as an IOS Enhanced Media Termination Point? Also, make sure you ALWAYS reset the trunk in CUCM. That will oftentimes clear out weird issues. I have learned that lesson the hard way. Matthew Berry *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* Wednesday, March 10, 2010 5:20 PM *To:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones Hello, the debug shows that the codec getting to the cme is g711 because the codec byte is 160 and still it disconnects On Thu, Mar 11, 2010 at 12:03 AM, Omotayo adefilabi...@gmail.com wrote: Hello, When i call from hq sccp phone to cme sip phone, it rings but when i pick up. it disconnects also when i call from cme sip phone to hq (sccp and sip) phone it rings on the hq phones when i pick t disconnects and contnues ringing on the sip phone i have a transcoder configured on the trunk Any one with a fix thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] BANDWIDTH SESSION
Hello All, Am working on Lab 6 Volume 2 When i issue the commnads below in order to ensure that 16Kps is used for calls between the UCM and CME for any particular call bandwidth total zone UCM 128 the call fails but when i remove bandwidth session UCM 16. Calls go through Anyone with an idea why i have this behaviour thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CM to GK BRQ behavior
Hello Scott, What is BRQ?? Am having siimilar issue how did you combat it. thanks On Wed, Jul 8, 2009 at 9:03 PM, Scott ODonnell scott.odonn...@gmail.comwrote: I'm seeing something strange in making calls from CM to CME via GK. I've enabled the BRQ service parameter in CM. I've included bandwidth total default 16 in my gk config and did a shut/no shut When I make calls from CM to CME the deb h225 asn1 shows (I think) that 128k is being requested. Am I missing something obivous here ? Currently all my calls get rejected from the GK and go via the HQ GW. If I remove the bandwidth command from the Gatekeeper config, the call works using g729. - Scott ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUCME SIP INTEGRATION WITH UC
Working on Volume 2 lab 6. integrating the cme with the unity connection. below is my config on the cme when i press the message button on extn 3002. it gives a bust tone Current configuration : 5763 bytes ! ! Last configuration change at 19:31:51 gmt Sat Mar 6 2010 ! NVRAM config last updated at 19:28:44 gmt Sat Mar 6 2010 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname BR2-RTR ! boot-start-marker boot system flash:c2800nm-adventerprisek9-mz.124-22.T.bin warm-reboot boot-end-marker ! logging message-counter syslog logging buffered 4096 ! no aaa new-model memory-size iomem 20 clock timezone gmt 0 clock summer-time CET recurring 1 Sun Apr 1:00 last Sun Oct 1:00 network-clock-participate wic 0 no network-clock-participate wic 1 network-clock-select 1 E1 0/0/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 10.10.202.1 10.10.202.49 ip dhcp excluded-address 10.10.202.70 10.10.202.254 ! ip dhcp pool br2 network 10.10.202.0 255.255.255.0 option 150 ip 10.10.202.1 default-router 10.10.202.1 ! ! 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 no supplementary-service sip moved-temporarily no supplementary-service sip refer sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 600 min 60 ! ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! ! ! ! ! ! ! ! ! ! ! ! voice register global mode cme source-address 10.10.110.3 port 5060 max-dn 3 max-pool 6 authenticate register mwi reg-e164 voicemail 3600 tftp-path flash: create profile sync 0002817099534509 ntp-server 10.10.100.2 mode unicast ! voice register dn 1 number 3002 call-forward b2bua busy 3600 call-forward b2bua mailbox 3002 call-forward b2bua noan 3600 timeout 12 name br2 phone 2 no-reg label br2 phone 2 mwi ! voice register pool 1 id mac .. type 7941 number 1 dn 1 dtmf-relay rtp-nte username 3002 password cisco codec g711ulaw ! ! voice translation-rule 1 rule 1 /3545623/ /3/ rule 2 /5623/ /3/ rule 3 /\+3545623/ /3/ ! voice translation-rule 2 rule 1 /^1#\(3...$\)/ /\1/ ! ! voice translation-profile GK translate called 2 ! voice translation-profile IN translate called 1 ! ! voice-card 0 ! ! ! ! ! archive log config hidekeys ! ! controller E1 0/0/0 pri-group timeslots 1-3,16 ! controller T1 0/1/0 channel-group 0 timeslots 1-24 ! ! ! ! ! interface Loopback0 ip address 10.10.110.3 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id UCM ipaddr 10.10.110.1 1719 h323-gateway voip h323-id UCME h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 10.10.110.3 ! interface FastEthernet0/0 no ip address shutdown duplex auto speed auto ! interface Service-Engine0/0 no ip address shutdown ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface FastEthernet0/3/0 switchport trunk native vlan 200 switchport mode trunk switchport voice vlan 400 ! interface FastEthernet0/3/1 shutdown ! interface FastEthernet0/3/2 shutdown ! interface FastEthernet0/3/3 shutdown ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn outgoing display-ie no cdp enable ! interface Serial0/1/0:0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/1/0:0.1 point-to-point ip address 10.10.112.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 102 ! interface Vlan1 no ip address ! interface Vlan200 ip address 10.10.102.1 255.255.255.0 ! interface Vlan400 ip address 10.10.202.1 255.255.255.0 ! router ospf 1 router-id 10.10.202.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! no ip http server no ip http secure-server ! ! ! ! ! ! control-plane ! ! ! voice-port 0/0/0:15 translation-profile incoming IN ! ccm-manager fax protocol cisco ! mgcp fax t38 ecm ! ! ! dial-peer voice 100 pots incoming called-number . direct-inward-dial ! dial-peer voice 200 voip huntstop max-conn 1 destination-pattern 3600 session protocol sipv2 session target ipv4:10.10.210.13 dtmf-relay rtp-nte codec g711ulaw ! dial-peer voice 300 voip destination-pattern [15]... session target ras dtmf-relay h245-a no vad voice-class codec 1 dial-peer voice 400 voip translation-profile incoming GK voice-class codec 1 incoming called-number 1#3... dtmf-relay h245-alphanumeric no vad ! ! gateway timer receive-rtp 1200 ! sip-ua mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp unsolicited ! ! telephony-service em logout 0:0 0:0 0:0 max-ephones 8
Re: [OSL | CCIE_Voice] Calls from hq to br2 thru gatekeeper
it fails before it gets to ring on the br2 phone. it gives calls can not be completed as dialled the calls gets to the br2 to gateway because i run the debug voice ccapi inout 2010/3/4 Roger Källberg roger.kallb...@cygate.se When does the call fail? If it fails just after you pick up then please try to add a voice-class codec list to your voip dial-peers that holds both g711 and g729. *Roger Källberg* -- *Från:* Omotayo [adefilabi...@gmail.com] *Skickat:* den 4 mars 2010 02:46 *Till:* OSL Group *Ämne:* Re: [OSL | CCIE_Voice] Calls from hq to br2 thru gatekeeper Hello, Now the call is getting to the cme but still says call can not be completed as dialled dial-peer voice 200 voip translation-profile incoming GK session target ras incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 300 voip destination-pattern [15]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric no vad ! ! gateway timer receive-rtp 1200 ! ! ! gatekeeper shutdown --More-- Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/cc_api_display_ie_subfields: cc_api_call_setup_ind_common: cisco-username=hq phone 2 - ccCallInfo IE subfields - cisco-ani=2123945002 cisco-anitype=0 cisco-aniplan=0 cisco-anipi=0 cisco-anisi=1 dest=1#3002 cisco-desttype=0 cisco-destplan=0 cisco-rdie= cisco-rdn= cisco-rdntype=-1 cisco-rdnplan=-1 cisco-rdnpi=-1 cisco-rdnsi=-1 cisco-redirectreason=-1 fwd_final_type =0 final_redirectNumber = hunt_group_timeout =0 Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/cc_api_call_setup_ind_common: Interface=0x4863B3F8, Call Info( Calling Number=2123945002,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), Called Number=1#3002(TON=Unknown, NPI=Unknown), Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Incoming Dial-peer=200, Progress Indication=NULL(0), Calling IE --More-- Present=TRUE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=188 Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/ccCheckClipClir: In: Calling Number=2123945002(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/ccCheckClipClir: Out: Calling Number=2123945002(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) Mar 4 01:44:01.035: //-1//CCAPI/cc_get_feature_vsa: Mar 4 01:44:01.035: :cc_get_feature_vsa malloc success Mar 4 01:44:01.035: //-1//CCAPI/cc_get_feature_vsa: Mar 4 01:44:01.035: cc_get_feature_vsa count is 1 Mar 4 01:44:01.035: //-1//CCAPI/cc_get_feature_vsa: Mar 4 01:44:01.035: :FEATURE_VSA attributes are: feature_name:0,feature_time:1248553792,feature_id:36 Mar 4 01:44:01.039: //188/004C02671500/CCAPI/cc_api_call_setup_ind_common: Set Up Event Sent; Call Info(Calling Number=2123945002(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), Called Number=1#3002(TON=Unknown, NPI=Unknown)) Mar 4 01:44:01.039: //188/004C02671500/CCAPI/cc_process_call_setup_ind: Event=0x499F5288 Mar 4 01:44:01.039: //-1//CCAPI/cc_setupind_match_search: Try with the demoted called number 1#3002 Mar 4 01:44:01.039: //188/004C02671500/CCAPI/ccCallSetContext: Context=0x4A5CF9EC Mar 4 01:44:01.039: //188/004C02671500/CCAPI/cc_process_call_setup_ind: CCAPI handed cid 188 with tag 200 to app _ManagedAppProcess_Default Mar 4 01:44:01.043: //188/004C02671500/CCAPI/ccCallProceeding: Progress Indication=NULL(0) Mar 4 01:44:01.043: //188/004C02671500/CCAPI/ccCallDisconnect: Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0) Mar 4 01:44:01.043: //188/004C02671500/CCAPI/ccCallDisconnect: Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1) Mar 4 01:44:01.047: //188/004C02671500/CCAPI/cc_api_get_transfer_info: Transfer Number Is Null Mar 4 01:44:01.055: //188/004C02671500/CCAPI/cc_api_call_disconnect_done: Disposition=0, Interface=0x4863B3F8, Tag=0x0, Call Id=188, Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0) Mar 4 01:44:01.055: //188/004C02671500/CCAPI/cc_api_call_disconnect_done: Call Disconnect Event Sent Mar 4 01:44:01.059: //-1//CCAPI/cc_free_feature_vsa: Mar 4 01:44:01.059: :cc_free_feature_vsa freeing 4A6B6B38 Mar 4 01:44:01.059: //-1//CCAPI/cc_free_feature_vsa: Mar 4 01:44:01.059: vsacount in free is 0 On Thu, Mar 4, 2010 at 2:16 AM, Omotayo adefilabi...@gmail.com wrote: Hello, Am working on lab 6 of volume 2 when i call from hq to br2, its gives your call can not be configured as dialled calls from br2 to hq works HQ-RTR#debug gatekeeper main 10 HQ-RTR# Mar 4
Re: [OSL | CCIE_Voice] SIP phone registered with CUCME
Hi, I have issues with the sip phone registering. it registers and keeps deregistering and registering intermittently On Thu, Mar 4, 2010 at 5:36 PM, iy...@nationwide.com wrote: Hi Guys, I was wondering if any one else had the same problem that I have with a SIP phone registered with CME CME version 7.0(1) on router 2821 running IOS 12.4(22) T3 Phone - 7962 running SIP load SIP42.8-5-3-4S The SIP phone is registered with the CME. I can make calls /receive calls, I call from a SCCP/SIP phone registered with CUCM or from a PSTN phone, call rings on the SIP phone and if I answer the call, everything is good. However if I don't answer the phone and if I dont hang up the phone I made the call from, the SIP phone keeps ringing on that line and never disconnects. I have to reset the phone by unplugging the ethernet and reboot the phone, Thanks Kalyan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Barge
Hello Otto, I want to make few clarification. If i have a centralized call processing system Hq phones need to barge into br1 phone. where do i need to configure the conferencing resouces ( hq router or br1 router) and which conferencing resources is used Also for transcoder, if br1 phones needs to access a server resources that is using g711 and br1 phones is coming with g729 cuz its going across the wan. where do i configure the transcoding resources- the hq router or br1 router thanks On Sun, Feb 21, 2010 at 8:10 PM, Otto Sanchez o...@ipexpert.com wrote: If you are barging br1phn2 the br1 conference resources are used, not the hq ones. Also make sure that br1phn2 has privacy off by the use of the privacy button or has privacy off in the device configuration, Finally, make sure that when cbarging from the shared line button, your hq phone config has the cbarge setting configured, On Sat, Feb 20, 2010 at 4:24 PM, Omotayo adefilabi...@gmail.com wrote: Hello, working on Volume 1 lab 8 IOS conference has been cnfigured on hq router when i tried with question 8.1 i used hq phones as bri phone of the question and vice versa on pressing the button on br1 phone 2 when in In Use Remote, i still see the barge softkey and it gives the message No Conference Bridge CBarge was enabled on the service parameter Below is a proof the conference bridge on the hq router is working HQ-RTR#sh sccp connections sess_idconn_idstype mode codec ripaddr rport sport 33557433 33554446 conf sendrecv g729b 192.168.3.1616386 16524 33557433 3355 conf sendrecv g729b 192.168.3.1218674 17838 33557433 33554442 conf sendrecv g711u 192.168.3.1828554 18950 Any ideas on what the issue is ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Barge
Hello Pulos, Thanks I am actually reading it at the moment. Guess am not clear On Wed, Mar 3, 2010 at 3:51 PM, Pulos, Greg gpu...@doc.gov wrote: The Cisco SRNDs for unified communications will tell you everything you need to know about how, where, why conferencing/transcoding/mtp resources are required. Please see the following link for more info on SRNDs for unified communications. http://www.cisco.com/iam/unified/ipt1/Using_SRND_Documents.htm greg -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo Sent: Wednesday, March 03, 2010 7:19 AM To: Otto Sanchez Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Barge Hello Otto, I want to make few clarification. If i have a centralized call processing system Hq phones need to barge into br1 phone. where do i need to configure the conferencing resouces ( hq router or br1 router) and which conferencing resources is used Also for transcoder, if br1 phones needs to access a server resources that is using g711 and br1 phones is coming with g729 cuz its going across the wan. where do i configure the transcoding resources- the hq router or br1 router thanks On Sun, Feb 21, 2010 at 8:10 PM, Otto Sanchez o...@ipexpert.com wrote: If you are barging br1phn2 the br1 conference resources are used, not the hq ones. Also make sure that br1phn2 has privacy off by the use of the privacy button or has privacy off in the device configuration, Finally, make sure that when cbarging from the shared line button, your hq phone config has the cbarge setting configured, On Sat, Feb 20, 2010 at 4:24 PM, Omotayo adefilabi...@gmail.com wrote: Hello, working on Volume 1 lab 8 IOS conference has been cnfigured on hq router when i tried with question 8.1 i used hq phones as bri phone of the question and vice versa on pressing the button on br1 phone 2 when in In Use Remote, i still see the barge softkey and it gives the message No Conference Bridge CBarge was enabled on the service parameter Below is a proof the conference bridge on the hq router is working HQ-RTR#sh sccp connections sess_idconn_idstype mode codec ripaddr rport sport 33557433 33554446 conf sendrecv g729b 192.168.3.16 16386 16524 33557433 3355 conf sendrecv g729b 192.168.3.12 18674 17838 33557433 33554442 conf sendrecv g711u 192.168.3.18 28554 18950 Any ideas on what the issue is ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calls from hq to br2 thru gatekeeper
Hello, Now the call is getting to the cme but still says call can not be completed as dialled dial-peer voice 200 voip translation-profile incoming GK session target ras incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 300 voip destination-pattern [15]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric no vad ! ! gateway timer receive-rtp 1200 ! ! ! gatekeeper shutdown --More-- Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/cc_api_display_ie_subfields: cc_api_call_setup_ind_common: cisco-username=hq phone 2 - ccCallInfo IE subfields - cisco-ani=2123945002 cisco-anitype=0 cisco-aniplan=0 cisco-anipi=0 cisco-anisi=1 dest=1#3002 cisco-desttype=0 cisco-destplan=0 cisco-rdie= cisco-rdn= cisco-rdntype=-1 cisco-rdnplan=-1 cisco-rdnpi=-1 cisco-rdnsi=-1 cisco-redirectreason=-1 fwd_final_type =0 final_redirectNumber = hunt_group_timeout =0 Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/cc_api_call_setup_ind_common: Interface=0x4863B3F8, Call Info( Calling Number=2123945002,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), Called Number=1#3002(TON=Unknown, NPI=Unknown), Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Incoming Dial-peer=200, Progress Indication=NULL(0), Calling IE --More-- Present=TRUE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=188 Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/ccCheckClipClir: In: Calling Number=2123945002(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/ccCheckClipClir: Out: Calling Number=2123945002(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) Mar 4 01:44:01.035: //-1//CCAPI/cc_get_feature_vsa: Mar 4 01:44:01.035: :cc_get_feature_vsa malloc success Mar 4 01:44:01.035: //-1//CCAPI/cc_get_feature_vsa: Mar 4 01:44:01.035: cc_get_feature_vsa count is 1 Mar 4 01:44:01.035: //-1//CCAPI/cc_get_feature_vsa: Mar 4 01:44:01.035: :FEATURE_VSA attributes are: feature_name:0,feature_time:1248553792,feature_id:36 Mar 4 01:44:01.039: //188/004C02671500/CCAPI/cc_api_call_setup_ind_common: Set Up Event Sent; Call Info(Calling Number=2123945002(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), Called Number=1#3002(TON=Unknown, NPI=Unknown)) Mar 4 01:44:01.039: //188/004C02671500/CCAPI/cc_process_call_setup_ind: Event=0x499F5288 Mar 4 01:44:01.039: //-1//CCAPI/cc_setupind_match_search: Try with the demoted called number 1#3002 Mar 4 01:44:01.039: //188/004C02671500/CCAPI/ccCallSetContext: Context=0x4A5CF9EC Mar 4 01:44:01.039: //188/004C02671500/CCAPI/cc_process_call_setup_ind: CCAPI handed cid 188 with tag 200 to app _ManagedAppProcess_Default Mar 4 01:44:01.043: //188/004C02671500/CCAPI/ccCallProceeding: Progress Indication=NULL(0) Mar 4 01:44:01.043: //188/004C02671500/CCAPI/ccCallDisconnect: Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0) Mar 4 01:44:01.043: //188/004C02671500/CCAPI/ccCallDisconnect: Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1) Mar 4 01:44:01.047: //188/004C02671500/CCAPI/cc_api_get_transfer_info: Transfer Number Is Null Mar 4 01:44:01.055: //188/004C02671500/CCAPI/cc_api_call_disconnect_done: Disposition=0, Interface=0x4863B3F8, Tag=0x0, Call Id=188, Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0) Mar 4 01:44:01.055: //188/004C02671500/CCAPI/cc_api_call_disconnect_done: Call Disconnect Event Sent Mar 4 01:44:01.059: //-1//CCAPI/cc_free_feature_vsa: Mar 4 01:44:01.059: :cc_free_feature_vsa freeing 4A6B6B38 Mar 4 01:44:01.059: //-1//CCAPI/cc_free_feature_vsa: Mar 4 01:44:01.059: vsacount in free is 0 On Thu, Mar 4, 2010 at 2:16 AM, Omotayo adefilabi...@gmail.com wrote: Hello, Am working on lab 6 of volume 2 when i call from hq to br2, its gives your call can not be configured as dialled calls from br2 to hq works HQ-RTR#debug gatekeeper main 10 HQ-RTR# Mar 4 01:10:08.527: ////GK/gk_process: got a TIMER event Mar 4 01:10:08.527: ////GK/gk_handle_timers Mar 4 01:10:08.527: ////GK/gk_handle_timers: managed timer expired 0x467B9F08 Mar 4 01:10:09.483: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Mar 4 01:10:09.483: ////GK/gk_rassrv_arq: arqp=0x4900F5C4,crv=0xA, answerCall=0 Mar 4 01:10:09.483: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Mar 4 01:10:09.483: //00B4D7AB0A00/00B4D7AB0A00/GK/gk_dns_query: No Name servers Mar 4 01:10:09.483: //00B4D7AB0A00/00B4D7AB0A00/GK/rassrv_get_addrinfo: (1
Re: [OSL | CCIE_Voice] gatekeeper call to cme
hello, i did all that On Tue, Mar 2, 2010 at 10:07 AM, Angel Perez gorr...@hotmail.com wrote: Hello: I suggest you the following: At gk-trunk: Uncheck wait for h245 capabilities and check fast start At cme site: You would need a transcoder for sip phone (just in case you have codec g711u at voice register pool) At gatekeeper: Delete all bandwith commands just to be sure your gk is not rejecting the call becouse insufficient bw, then deb gatekeeper main 10, also deb h225 asn1 could be helpfull (and verbose) hth -- Date: Tue, 2 Mar 2010 08:26:27 +0100 From: adefilabi...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] gatekeeper call to cme Hello, Anyone with a clue? Regards On Sat, Feb 27, 2010 at 10:33 PM, Omotayo adefilabi...@gmail.com wrote: Hello, working on Vol 1 lab 4 Calls from hq and Bri to br2 works with all sccp phones with sip phone( i used xlite) at br2 when the sip phone calls hq/br1 Calls continues to ring on the sip phone while it disconnects on hq/br1 phone when Hq/br1 calls sip phone it rings on the sipphones but as soon as the sip phone picjs it gives a busy tone on hq/br1 phone When i uncheck the Enable Inbound faststart Calls from sip to hq/br1 disconnects after 10sec of connection calls from hq/br1 to sip disconnects afters 2 secs of connection i have transcoder on hq phones and the trunk the trunk has a device pools that does g729r8 to all regions Below is the configuration on the hq gateway and br2 gateway Q-RTR(config)#do sh run Building configuration... Current configuration : 4210 bytes ! ! Last configuration change at 09:40:44 pst Sat Feb 27 2010 ! NVRAM config last updated at 12:25:10 pst Sat Feb 27 2010 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ-RTR ! boot-start-marker warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 clock timezone pst -8 network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! vtp domain home.com vtp mode transparent archive log config hidekeys ! ! ! ! controller T1 0/0/0 framing esf linecode ami pri-group timeslots 1-3,24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ip rsvp bandwidth ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay fair-queue 64 256 36 frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ip rsvp bandwidth 64 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! ! ip http server no ip http secure-server ! ! ! ! ! ! ! control-plane ! rmon event 3 log trap AutoQoS description AutoQoS SNMP traps for Voice Drops owner AutoQoS ! ! voice-port 0/0/0:23 ! ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ! mgcp mgcp call-agent 10.10.210.11 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface FastEthernet0/0.20 mgcp bind media source-interface FastEthernet0/0.20 ! mgcp profile default ! sccp local FastEthernet0/0.20 sccp ccm 10.10.210.11 identifier 1 version 5.0.1 sccp ccm 10.10.210.10 identifier 2 version 5.0.1 sccp ! sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate
Re: [OSL | CCIE_Voice] gatekeeper call to cme
Hello, Anyone with a clue? Regards On Sat, Feb 27, 2010 at 10:33 PM, Omotayo adefilabi...@gmail.com wrote: Hello, working on Vol 1 lab 4 Calls from hq and Bri to br2 works with all sccp phones with sip phone( i used xlite) at br2 when the sip phone calls hq/br1 Calls continues to ring on the sip phone while it disconnects on hq/br1 phone when Hq/br1 calls sip phone it rings on the sipphones but as soon as the sip phone picjs it gives a busy tone on hq/br1 phone When i uncheck the Enable Inbound faststart Calls from sip to hq/br1 disconnects after 10sec of connection calls from hq/br1 to sip disconnects afters 2 secs of connection i have transcoder on hq phones and the trunk the trunk has a device pools that does g729r8 to all regions Below is the configuration on the hq gateway and br2 gateway Q-RTR(config)#do sh run Building configuration... Current configuration : 4210 bytes ! ! Last configuration change at 09:40:44 pst Sat Feb 27 2010 ! NVRAM config last updated at 12:25:10 pst Sat Feb 27 2010 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ-RTR ! boot-start-marker warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 clock timezone pst -8 network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! vtp domain home.com vtp mode transparent archive log config hidekeys ! ! ! ! controller T1 0/0/0 framing esf linecode ami pri-group timeslots 1-3,24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ip rsvp bandwidth ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay fair-queue 64 256 36 frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ip rsvp bandwidth 64 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! ! ip http server no ip http secure-server ! ! ! ! ! ! ! control-plane ! rmon event 3 log trap AutoQoS description AutoQoS SNMP traps for Voice Drops owner AutoQoS ! ! voice-port 0/0/0:23 ! ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ! mgcp mgcp call-agent 10.10.210.11 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface FastEthernet0/0.20 mgcp bind media source-interface FastEthernet0/0.20 ! mgcp profile default ! sccp local FastEthernet0/0.20 sccp ccm 10.10.210.11 identifier 1 version 5.0.1 sccp ccm 10.10.210.10 identifier 2 version 5.0.1 sccp ! sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 2 register hq-transcode associate profile 1 register hq-mtp ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 2 associate application SCCP ! dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 2 associate application SCCP ! ! ! ! ! gatekeeper zone local PL ipexpert.com zone prefix PL 1... gw-priority 10 gk-trunk_2 zone prefix PL 1... gw-priority 9 gk-trunk_1 zone prefix PL 5... gw-priority 10 gk-trunk_2 zone prefix PL 5... gw-priority 9 gk-trunk_1 no shutdown ! ! line con 0 exec-timeout 0 0 logging synchronous line aux 0 line vty 0
[OSL | CCIE_Voice] bacd issue
Hello, Still on the BACD, when i dial 32143007 from PSTN or 3007 for internal phones. i get the Unknown number tone Kindly help check the config and see whatz missing thanks BR2-RTR#sh run Building configuration... Current configuration : 8096 bytes ! ! No configuration change since last restart ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname BR2-RTR ! boot-start-marker warm-reboot boot-end-marker ! logging message-counter syslog logging buffered 4096 ! no aaa new-model memory-size iomem 20 clock timezone hkt 8 network-clock-participate wic 0 no network-clock-participate wic 1 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 10.10.202.1 10.10.202.9 ip dhcp excluded-address 10.10.202.31 10.10.202.254 ! ip dhcp pool SiteC network 10.10.202.0 255.255.255.0 default-router 10.10.202.1 option 150 ip 10.10.202.1 ! ! 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 h323 h225 listen-port 1820 no call service stop sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 300 min 60 ! ! ! voice class codec 1 codec preference 1 g729r8 bytes 20 codec preference 2 g711ulaw bytes 160 ! ! ! voice class h323 1 ! ! ! ! ! ! ! ! ! ! voice register global mode cme source-address 10.10.110.3 port 5060 max-dn 3 max-pool 3 authenticate register voicemail 3600 tftp-path flash: ntp-server 10.10.100.2 mode unicast ! voice register dn 1 number 3004 call-forward b2bua busy 3600 call-forward b2bua mailbox 3004 call-forward b2bua noan 360 timeout 15 name SiteB Phone 4 no-reg label SiteC Phone 4 mwi ! voice register pool 1 id mac .. type 7960 number 1 dn 1 dtmf-relay rtp-nte username 3004 password cisco codec g711ulaw no vad ! voice hunt-group 1 parallel list 3002,3006 pilot 3210 ! ! ! voice translation-rule 1 rule 1 /3432143/ /3/ rule 2 /32143/ /3/ rule 3 /\+3432143/ /3/ ! voice translation-rule 2 rule 1 /^\(3...\)$/ /+343214\1/ ! ! voice translation-profile IN translate called 1 ! voice translation-profile OUT translate calling 2 ! ! voice-card 0 no dspfarm dsp services dspfarm ! ! application service queue flash:bacdprompts/app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt3 3002 param queue-manager-debugs 1 param aa-hunt2 3210 param number-of-hunt-grps 2 ! service aa flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 param number-of-hunt-grps 2 param handoff-string aa param dial-by-extension-option 3 paramspace english language en param max-time-vm-retry 2 param aa-pilot 3007 paramspace english location flash:bacdprompts/ param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param voice-mail 3600 param max-time-call-retry 700 param service-name queue ! ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller E1 0/0/0 pri-group timeslots 1-3,16 ! controller T1 0/1/0 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! ! ! ! interface Loopback0 ip address 10.10.110.3 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id PL ipaddr 10.10.110.1 1719 h323-gateway voip h323-id BR2-RTR h323-gateway voip tech-prefix 3 ! interface FastEthernet0/0 no ip address shutdown duplex auto speed auto ! interface Service-Engine0/0 ip unnumbered Vlan400 service-module ip address 10.10.202.2 255.255.255.0 service-module ip default-gateway 10.10.202.1 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface FastEthernet0/3/0 switchport trunk native vlan 200 switchport mode trunk switchport voice vlan 400 ! interface FastEthernet0/3/1 switchport trunk native vlan 200 switchport mode trunk switchport voice vlan 400 ! interface FastEthernet0/3/2 shutdown ! interface FastEthernet0/3/3 shutdown ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn bchan-number-order ascending isdn outgoing display-ie no cdp enable ! interface Serial0/1/0:0 no ip address encapsulation frame-relay IETF frame-relay lmi-type ansi ! interface Serial0/1/0:0.1 point-to-point ip address 10.10.112.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 102 ! interface Vlan1 no ip address ! interface Vlan200 ip address 10.10.102.1 255.255.255.0 ! interface Vlan400 ip address 10.10.202.1 255.255.255.0 ! router ospf 1 router-id 10.10.202.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ip route 10.10.202.2 255.255.255.255 Service-Engine0/0 ip http server no ip http secure-server ip http path
[OSL | CCIE_Voice] gatekeeper call to cme
Hello, working on Vol 1 lab 4 Calls from hq and Bri to br2 works with all sccp phones with sip phone( i used xlite) at br2 when the sip phone calls hq/br1 Calls continues to ring on the sip phone while it disconnects on hq/br1 phone when Hq/br1 calls sip phone it rings on the sipphones but as soon as the sip phone picjs it gives a busy tone on hq/br1 phone When i uncheck the Enable Inbound faststart Calls from sip to hq/br1 disconnects after 10sec of connection calls from hq/br1 to sip disconnects afters 2 secs of connection i have transcoder on hq phones and the trunk the trunk has a device pools that does g729r8 to all regions Below is the configuration on the hq gateway and br2 gateway Q-RTR(config)#do sh run Building configuration... Current configuration : 4210 bytes ! ! Last configuration change at 09:40:44 pst Sat Feb 27 2010 ! NVRAM config last updated at 12:25:10 pst Sat Feb 27 2010 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ-RTR ! boot-start-marker warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 clock timezone pst -8 network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! vtp domain home.com vtp mode transparent archive log config hidekeys ! ! ! ! controller T1 0/0/0 framing esf linecode ami pri-group timeslots 1-3,24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ip rsvp bandwidth ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay fair-queue 64 256 36 frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ip rsvp bandwidth 64 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! ! ip http server no ip http secure-server ! ! ! ! ! ! ! control-plane ! rmon event 3 log trap AutoQoS description AutoQoS SNMP traps for Voice Drops owner AutoQoS ! ! voice-port 0/0/0:23 ! ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ! mgcp mgcp call-agent 10.10.210.11 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface FastEthernet0/0.20 mgcp bind media source-interface FastEthernet0/0.20 ! mgcp profile default ! sccp local FastEthernet0/0.20 sccp ccm 10.10.210.11 identifier 1 version 5.0.1 sccp ccm 10.10.210.10 identifier 2 version 5.0.1 sccp ! sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 2 register hq-transcode associate profile 1 register hq-mtp ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 2 associate application SCCP ! dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 2 associate application SCCP ! ! ! ! ! gatekeeper zone local PL ipexpert.com zone prefix PL 1... gw-priority 10 gk-trunk_2 zone prefix PL 1... gw-priority 9 gk-trunk_1 zone prefix PL 5... gw-priority 10 gk-trunk_2 zone prefix PL 5... gw-priority 9 gk-trunk_1 no shutdown ! ! line con 0 exec-timeout 0 0 logging synchronous line aux 0 line vty 0 4 exec-timeout 0 0 privilege level 15 logging synchronous no login length 0 transport preferred none transport input telnet line vty 5 15 exec-timeout 0 0 privilege level 15 logging synchronous no login transport preferred none transport input telnet ! scheduler allocate 2 1000 ntp clock-period 17179862 ntp server 10.10.100.2 ! end
[OSL | CCIE_Voice] BACD
Hello, Working on the BACD for volume 1 in my lab i have the following. when i dial 1 i get the messgae you have entered an invalid entry when i dial 2 i get thereis no mail box associated with ths extension Only dial0ng by extension and operator works what could be the issue voice service voip allow-connection h t s telephony-service moh music-on-hold.au application service queue flash:app-b-acd-2.1.2.2.tcl param number-of-hunt-grps 2 param aa-hunt1 888 param aa-hunt2 999 param aa-hunt10 100 param queue-len 15 param queue-manager-debugs 1 service aa flash:app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 paramspace english language en paramspace english location flash: param service-name queue param handoff-string aa param aa-pilot 800 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 3 param dial-by-extension-option 3 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 90 param max-time-vm-retry 2 param voice-mail 333 dial-peer voice 800 voip service aa destination-pattern 800 session target ipv4:172.168.10.1 incoming called-number 800 dtmf-relay h245-alphanumeric codec g711ulaw no vad ephone-hunt 1 sequential pilot 999 list 104, 103 timeout 10, 10 ! ! ephone-hunt 2 peer pilot 888 list 109, 105, 107, 111 ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Barge
Hello, working on Volume 1 lab 8 IOS conference has been cnfigured on hq router when i tried with question 8.1 i used hq phones as bri phone of the question and vice versa on pressing the button on br1 phone 2 when in In Use Remote, i still see the barge softkey and it gives the message No Conference Bridge CBarge was enabled on the service parameter Below is a proof the conference bridge on the hq router is working HQ-RTR#sh sccp connections sess_idconn_idstype mode codec ripaddr rport sport 33557433 33554446 conf sendrecv g729b 192.168.3.1616386 16524 33557433 3355 conf sendrecv g729b 192.168.3.1218674 17838 33557433 33554442 conf sendrecv g711u 192.168.3.1828554 18950 Any ideas on what the issue is ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Number-type
Hello, Working on Vol 1 lab 5.4 i tried to configure nubering-typer subscriber under the dial-peer for the local calls on the h323 gateway. but call failed with number type mismatch i guess this is because of the configuration on the pstn gateway right? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Already seen the new Vol1 en 2 labs?
Hello, Which is the new vol workbook and PG because i think i still have the former in my memeber account any one with the new ones?? On Sun, Jan 31, 2010 at 5:25 PM, Robert McGhee bobwmcg...@verizon.netwrote: The new WB’s look very good, any idea when the solutions will be available as well? Thank you for the updates…. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Wayne Lawson *Sent:* Sunday, January 31, 2010 9:19 AM *To:* Bas Janssen *Cc:* ccie_voice@onlinestudylist.com; Ryan Barnum *Subject:* Re: [OSL | CCIE_Voice] Already seen the new Vol1 en 2 labs? Bas - I've copied my support team on this. I know they were swamped ALL WEEKEND with all of the massive updates / files that needed to be updated. I anticipate this being totally resolve today (for everyone, all tracks). Youre gonna LOVE the new material - it's amazing!! Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 101 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com. On Jan 31, 2010, at 6:14 AM, Bas Janssen basmj...@msn.com wrote: Hi, I am curious to know if somebody already received the additional Vol1 en 2 labs in the download section. I haven't seen them yet. Regards, Bas -- Hotmail: Powerful Free email with security by Microsoft. Get it now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Proctor labs down
I can access it its up On Fri, Jan 29, 2010 at 2:10 PM, Arun Kumar arunv...@gmail.com wrote: Hi All, Is anyone else able to access ? I can't access web , no rack via telnet no vpn ? I can't access my session. Thanks Arun ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UCCX clock set
Hello, Am working on Lab 2 how do i synchronize the uccx time to that of the callmanager or the ntp(10.10.100.2) thanks for the anticipated help rgd ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX clock set
Hello, IN addition, the time on the control center is different from that on the server time thanks On Thu, Jan 14, 2010 at 4:03 PM, Omotayo adefilabi...@gmail.com wrote: Hello, Am working on Lab 2 how do i synchronize the uccx time to that of the callmanager or the ntp(10.10.100.2) thanks for the anticipated help rgd ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX clock set
hello, i get that part. the issue i have is that on the control center. the time is January 14, 2010 10:27:51 AM EST while on the server its is January 14, 2010 7:27:51 AM EST the zone set on the server is gmt-8 On Thu, Jan 14, 2010 at 4:16 PM, Pulos, Greg gpu...@doc.gov wrote: Use windows time or xntp provided by Cisco. Please see the link below for more info on time servers for UCXX. http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_5_0/maintenance/admin/crs501ag.pdf Page 4-12; Modifying NTP Configuration Thank you. greg -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo Sent: Thursday, January 14, 2010 10:03 AM To: OSL Group Subject: [OSL | CCIE_Voice] UCCX clock set Hello, Am working on Lab 2 how do i synchronize the uccx time to that of the callmanager or the ntp(10.10.100.2) thanks for the anticipated help rgd ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UCM AND CUE INTEGRATION
Hello All, Am working on lab 4 i have integrated the UCM to CUE. Everything shows registered end CUE# show ccn status ccm-manager JTAPI Subsystem is currently registered with Call Manager: 10.10.210.10 JTAPI Version: 7.0(1.1000) -1 Release when i dial the pilot number 3600, i get a busy tone when i dial the CTI Port, it gives Not Enough Bandwidth. I suspect its the location configurtion affecting it . i have 48 configured for the branch 2 location has requested by the question to allow just 2 g729 calls between locations When i changed the bandwidth to 96. it worked Anyone with an explanation for this behavouir REgards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MLPP and traffic shaping
Hello, i have configured MLP LFI between hq and br1 router. i reloaded the routers but i can not ping across( this i can do prior to the configuration) with show policy-map i have the following output HQ-RTR#sh policy-map Interface Virtual-Access1 Virtual-Access1 Service-policy output: policy Service policy policy is in suspended mode HQ-RTR#ping 10.10.201.1 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.201.1, timeout is 2 seconds: . Success rate is 0 percent (0/5) HQ-RTR# On the br1 router, i have the following BR1-RTR#sh policy-map Interface Virtual-Access1 Virtual-Access1 Service-policy output: policy queue stats for all priority classes: queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 Class-map: media (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp ef (46) 0 packets, 0 bytes 5 minute rate 0 bps Priority: 33% (126 kbps), burst bytes 3150, b/w exceed drops: 0 Class-map: control (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp cs3 (24) 0 packets, 0 bytes 5 minute rate 0 bps Match: ip dscp af31 (26) 0 packets, 0 bytes 5 minute rate 0 bps Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 bandwidth 5% (19 kbps) Class-map: class-default (match-any) 254 packets, 6482 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops/flowdrops) 250/3/0/3 (pkts output/bytes output) 251/6407 Fair-queue: per-flow queue limit 16 BR1-RTR# ping 10.10.200.3 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.200.3, timeout is 2 seconds: . Success rate is 0 percent (0/5) BR1-RTR# When I set the service-policy output , I get an error message Class Based Weighted Fair Queueing will be applied only to the Virtual-Access interfaces associated with an MLP bundle. Any ideas on what is wrong thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MLPP and traffic shaping
! interface FastEthernet1/1 ! interface FastEthernet1/2 ! interface FastEthernet1/3 ! interface FastEthernet1/4 ! interface FastEthernet1/5 ! interface FastEthernet1/6 ! interface FastEthernet1/7 ! interface FastEthernet1/8 ! interface FastEthernet1/9 ! interface FastEthernet1/10 ! interface FastEthernet1/11 ! interface FastEthernet1/12 ! interface FastEthernet1/13 ! interface FastEthernet1/14 ! interface FastEthernet1/15 ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.2 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output POLICY-CHECK ! interface Vlan1 no ip address ! interface Vlan130 ip address 10.10.101.1 255.255.255.0 ! interface Vlan240 ip address 10.10.201.1 255.255.255.0 ip helper-address 10.10.210.10 ! router ospf 1 router-id 10.10.101.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd no ip http server no ip http secure-server ! ! ! ! map-class frame-relay traffic frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 ! ! ! ! ! ! tftp-server flash:moh-on-hold.au ! control-plane ! ! ! voice-port 0/0/0:23 On Tue, Jan 5, 2010 at 12:14 PM, Graham Hopkins ghopk...@wolf-rock.co.ukwrote: Can you post the interface configurations. I've had some issues in this area and you do need to ensure all the templates and policies line up. For example I'm still investigating a situation where I get two cloned virtual-acess interfaces and only one has the service policy applied ! interface Virtual-Access1 bandwidth 768 ip address 10.10.112.1 255.255.255.0 end HQ-RTR#sh run int virtual-access 3 Building configuration... Current configuration : 117 bytes ! interface Virtual-Access3 bandwidth 768 ip address 10.10.112.1 255.255.255.0 service-policy output 768kbps end Regards Graham Hopkins On 5 Jan 2010, at 09:23, Omotayo wrote: Hello, i have configured MLP LFI between hq and br1 router. i reloaded the routers but i can not ping across( this i can do prior to the configuration) with show policy-map i have the following output HQ-RTR#sh policy-map Interface Virtual-Access1 Virtual-Access1 Service-policy output: policy Service policy policy is in suspended mode HQ-RTR#ping 10.10.201.1 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.201.1, timeout is 2 seconds: . Success rate is 0 percent (0/5) HQ-RTR# On the br1 router, i have the following BR1-RTR#sh policy-map Interface Virtual-Access1 Virtual-Access1 Service-policy output: policy queue stats for all priority classes: queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 Class-map: media (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp ef (46) 0 packets, 0 bytes 5 minute rate 0 bps Priority: 33% (126 kbps), burst bytes 3150, b/w exceed drops: 0 Class-map: control (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp cs3 (24) 0 packets, 0 bytes 5 minute rate 0 bps Match: ip dscp af31 (26) 0 packets, 0 bytes 5 minute rate 0 bps Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 bandwidth 5% (19 kbps) Class-map: class-default (match-any) 254 packets, 6482 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops/flowdrops) 250/3/0/3 (pkts output/bytes output) 251/6407 Fair-queue: per-flow queue limit 16 BR1-RTR# ping 10.10.200.3 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.200.3, timeout is 2 seconds: . Success rate is 0 percent (0/5) BR1-RTR# When I set the service-policy output , I get an error message Class Based Weighted Fair Queueing will be applied only to the Virtual-Access interfaces associated with an MLP bundle. Any ideas on what is wrong thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ILT Lab 1 Question 4.4 - No default technology prefix?!
Hello, for me i can call from HQ to br2 but can not call from br2 to hq i always get a unknwn number tone What do you think is missing thanks Below is the output when calling from br2 to hq Also when i removed the bandwidth session 16, it goes through but you wont hear the callers'seech and gets disconnected after a while with a reorder tone HQ-RTR# Dec 22 01:40:12.129: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Dec 22 01:40:12.129: ////GK/gk_rassrv_arq: arqp=0x48DAF694,crv=0x3C, answerCall=0 Dec 22 01:40:12.129: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Dec 22 01:40:12.129: //C7A4073280DD/C7A53F8280DF/GK/gk_dns_query: No Name servers Dec 22 01:40:12.129: //C7A4073280DD/C7A53F8280DF/GK/rassrv_get_addrinfo: (1#12123945001) Matched tech-prefix 1# Dec 22 01:40:12.129: //C7A4073280DD/C7A53F8280DF/GK/rassrv_get_addrinfo: (1#12123945001) unresolved zone prefix, using source zone HQ Dec 22 01:40:12.129: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 3 Dec 22 01:40:12.129: //C7A4073280DD/C7A53F8280DF/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x48C88C44 Dec 22 01:40:12.129: //C7A4073280DD/C7A53F8280DF/GK/rassrv_arq_select_viazone: matched zone is HQ, and z_outvian HQ-RTR#amelen=0 Dec 22 01:40:12.129: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 3 Dec 22 01:40:12.153: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Dec 22 01:40:12.153: ////GK/gk_rassrv_arq: arqp=0x48DAF694,crv=0x803C, answerCall=1 Dec 22 01:40:12.153: //C7A4073280DD/C7A53F8280DF/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Dec 22 01:40:12.177: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Dec 22 01:40:12.177: ////GK/gk_rassrv_arq: arqp=0x48D9B850,crv=0x803D, answerCall=1 Dec 22 01:40:12.177: //C7A4073280DD/C7A53F8280DF/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Dec 22 01:40:12.189: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Dec 22 01:40:12.233: ////GK/gk_process: got a TIMER event Thanks for the anticipated support On Tue, Nov 17, 2009 at 12:35 AM, Vik Malhi vma...@ipexpert.com wrote: Q4.4 is dealing with registration to GK. The dial-peer is created in the call routing section: On the BR2-RTR: *voice translation-rule 15 rule 1 /^5...$/ /1#1212394\0/ rule 2 /^1...$/ /1#1617863\0/ ! voice translation-profile GK-OUT translate called 15 ! dial-peer voice 15 voip destination-pattern [15]...$ session target ras no vad dtmf-relay h245-alphanumberic translation-profile out GK-OUT * -- Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities http://www.ipexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Nara Shikamaru shikam...@kagadis.com *Date: *Mon, 16 Nov 2009 15:44:45 -0700 *To: *Kumar, Narinder narinder.ku...@uxcg.com.au *Cc: *OSL Group ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] ILT Lab 1 Question 4.4 - No default technology prefix?! This config is puzzling. There should at least be a dial peer on BR2 pointing to the gatekeep with *session target ras*. The only dial-peers that seem to be pointing traffic to the 1 and 5 patterns are; dial-peer voice 15 voip destination-pattern [15]...$ voice-class codec 1 session protocol sipv2 session target ipv4:10.10.210.11 incoming called-number . no vad ! dial-peer voice 16 voip preference 1 destination-pattern [15]...$ voice-class codec 1 session protocol sipv2 session target ipv4:10.10.210.10 no vad On Mon, Nov 16, 2009 at 3:11 PM, Kumar, Narinder narinder.ku...@uxcg.com.au wrote: I don’t remember what is the exact requirements of Lab 4.4. Are you sending any tech prefix from ur gateway’s when they are registering with the gatekeeper ? Not sure what will happen if the gateways are registering with gatekeeper with a different tech prefix and GK is using the default tech prefix , which one takes priority or what happens. I need to read up on the gatekeepers again. *From:* ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Nara Shikamaru *Sent:* Tuesday, 17 November 2009 8:59 AM *To:* OSL Group *Subject:* [OSL | CCIE_Voice] ILT Lab 1 Question 4.4 - No default technology
Re: [OSL | CCIE_Voice] ILT Lab 1 Question 4.4 - No default technology prefix?!
The files available on the member account dont have the final config files just the initials On Tue, Nov 17, 2009 at 4:10 PM, Vik Malhi vma...@ipexpert.com wrote: The final configuration files should not be used for any lab since we change labs and solutions more regularly than we can update final configuration files. Please ignore the final configuration button on PL from now on. The solution doc for ILT lab 1 is accurate and does not show default tech prefix being set. -- Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities http://www.ipexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Nara Shikamaru shikam...@kagadis.com *Date: *Mon, 16 Nov 2009 14:58:50 -0700 *To: *OSL Group ccie_voice@onlinestudylist.com *Subject: *[OSL | CCIE_Voice] ILT Lab 1 Question 4.4 - No default technology prefix?! I'm very confused. In this question, we are told very explicitly in bulltet 3 that we are not allowed to use the default technology prefix sytnax. But, in the final configuration file for the gatekeeper it's actually used; *gatekeeper zone local US ipexpert.com http://ipexpert.com zone local Spain ipexpert.com http://ipexpert.com zone remote PSTN-WAN ipexpert.com http://ipexpert.com 10.10.100.2 1719 zone prefix Spain 34* zone prefix PSTN-WAN 91* gw-type-prefix 1#* default-technology no shutdown endpoint resource-threshold endpoint max-calls h323id gk-trunk_2 1 * Can someone help me understand? Am I seeing this correctly? -- -Shikamaru -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MLP LFI
Hello, Working on Proctor lab. i need to configure MLP LFI between HQ and BR1 with link speed of 384kbps. I have configured but link wont come up i did a shw ppp multilink but got No active bundles No inactive multilink interfaces Dont know why this is. Below are my configs HQ-RTR#show frame-relay pvc 201 PVC Statistics for interface Serial0/0/1:0 (Frame Relay DTE) DLCI = 201, DLCI USAGE = LOCAL, PVC STATUS = ACTIVE, INTERFACE = Serial0/0/1:0.1 input pkts 16output pkts 216 in bytes 5776 out bytes 11256 dropped pkts 0 in pkts dropped 0 out pkts dropped 0out bytes dropped 0 in FECN pkts 0 in BECN pkts 0 out FECN pkts 0 out BECN pkts 0 in DE pkts 0 out DE pkts 0 out bcast pkts 16out bcast bytes 5456 5 minute input rate 0 bits/sec, 0 packets/sec 5 minute output rate 0 bits/sec, 0 packets/sec pvc create time 00:16:49, last time pvc status changed 00:16:21 Bound to Virtual-Access1 (down, cloned from Virtual-Template200) cir 364800bc 3648 be 0 byte limit 456interval 10 mincir 364800byte increment 456 Adaptive Shaping none pkts 216 bytes 11256 pkts delayed 0 bytes delayed 0 shaping inactive traffic shaping drops 0 Queueing strategy: fifo Output queue 0/40, 0 drop, 0 dequeued HQ-RTR#sh HQ-RTR#show ppp mul HQ-RTR#show ppp multilink No active bundles No inactive multilink interfaces Q-RTR# HQ-RTR# HQ-RTR# HQ-RTR# HQ-RTR# HQ-RTR#sh run Building configuration... Current configuration : 4382 bytes ! ! Last configuration change at 21:17:23 UTC Thu Dec 17 2009 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ-RTR ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs-mz.124-20.T1.bin warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! vtp mode transparent archive log config hidekeys ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! class-map match-any CONTROL match ip dscp cs3 match ip dscp af31 class-map match-any RTP match ip dscp ef ! ! policy-map POLICY-CHECK class RTP priority percent 33 compress header ip rtp class CONTROL bandwidth percent 5 class class-default fair-queue ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay no fair-queue frame-relay traffic-shaping frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template200 class traffic-shape ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.1 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output POLICY-CHECK ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! ! no ip http server ip http authentication local no ip http secure-server ! ! map-class frame-relay traffic-shape frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 ! ! ! ! ! ! control-plane ! ! ! voice-port 0/0/0:23 ! ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ccm-manager music-on-hold ! mgcp mgcp call-agent 10.10.210.11 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface FastEthernet0/0.20 mgcp bind media source-interface FastEthernet0/0.20 ! mgcp profile default ! sccp local FastEthernet0/0.20 sccp ccm 10.10.210.11 identifier 1 version 5.0.1 sccp ccm 10.10.210.10 identifier 2 version 5.0.1 sccp !
Re: [OSL | CCIE_Voice] MLP LFI
Hello, Also i can not ping across from HQ to 10.10.202.1 On Thu, Dec 17, 2009 at 10:28 PM, Omotayo adefilabi...@gmail.com wrote: Hello, Working on Proctor lab. i need to configure MLP LFI between HQ and BR1 with link speed of 384kbps. I have configured but link wont come up i did a shw ppp multilink but got No active bundles No inactive multilink interfaces Dont know why this is. Below are my configs HQ-RTR#show frame-relay pvc 201 PVC Statistics for interface Serial0/0/1:0 (Frame Relay DTE) DLCI = 201, DLCI USAGE = LOCAL, PVC STATUS = ACTIVE, INTERFACE = Serial0/0/1:0.1 input pkts 16output pkts 216 in bytes 5776 out bytes 11256 dropped pkts 0 in pkts dropped 0 out pkts dropped 0out bytes dropped 0 in FECN pkts 0 in BECN pkts 0 out FECN pkts 0 out BECN pkts 0 in DE pkts 0 out DE pkts 0 out bcast pkts 16out bcast bytes 5456 5 minute input rate 0 bits/sec, 0 packets/sec 5 minute output rate 0 bits/sec, 0 packets/sec pvc create time 00:16:49, last time pvc status changed 00:16:21 Bound to Virtual-Access1 (down, cloned from Virtual-Template200) cir 364800bc 3648 be 0 byte limit 456interval 10 mincir 364800byte increment 456 Adaptive Shaping none pkts 216 bytes 11256 pkts delayed 0 bytes delayed 0 shaping inactive traffic shaping drops 0 Queueing strategy: fifo Output queue 0/40, 0 drop, 0 dequeued HQ-RTR#sh HQ-RTR#show ppp mul HQ-RTR#show ppp multilink No active bundles No inactive multilink interfaces Q-RTR# HQ-RTR# HQ-RTR# HQ-RTR# HQ-RTR# HQ-RTR#sh run Building configuration... Current configuration : 4382 bytes ! ! Last configuration change at 21:17:23 UTC Thu Dec 17 2009 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ-RTR ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs-mz.124-20.T1.bin warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! vtp mode transparent archive log config hidekeys ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! class-map match-any CONTROL match ip dscp cs3 match ip dscp af31 class-map match-any RTP match ip dscp ef ! ! policy-map POLICY-CHECK class RTP priority percent 33 compress header ip rtp class CONTROL bandwidth percent 5 class class-default fair-queue ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay no fair-queue frame-relay traffic-shaping frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template200 class traffic-shape ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.1 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output POLICY-CHECK ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! ! no ip http server ip http authentication local no ip http secure-server ! ! map-class frame-relay traffic-shape frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 ! ! ! ! ! ! control-plane ! ! ! voice-port 0/0/0:23 ! ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ccm-manager music-on-hold ! mgcp mgcp
Re: [OSL | CCIE_Voice] MLP LFI
Hello, Not to worry. i figured it out i had a DLCI mismatch On Thu, Dec 17, 2009 at 10:28 PM, Omotayo adefilabi...@gmail.com wrote: Hello, Working on Proctor lab. i need to configure MLP LFI between HQ and BR1 with link speed of 384kbps. I have configured but link wont come up i did a shw ppp multilink but got No active bundles No inactive multilink interfaces Dont know why this is. Below are my configs HQ-RTR#show frame-relay pvc 201 PVC Statistics for interface Serial0/0/1:0 (Frame Relay DTE) DLCI = 201, DLCI USAGE = LOCAL, PVC STATUS = ACTIVE, INTERFACE = Serial0/0/1:0.1 input pkts 16output pkts 216 in bytes 5776 out bytes 11256 dropped pkts 0 in pkts dropped 0 out pkts dropped 0out bytes dropped 0 in FECN pkts 0 in BECN pkts 0 out FECN pkts 0 out BECN pkts 0 in DE pkts 0 out DE pkts 0 out bcast pkts 16out bcast bytes 5456 5 minute input rate 0 bits/sec, 0 packets/sec 5 minute output rate 0 bits/sec, 0 packets/sec pvc create time 00:16:49, last time pvc status changed 00:16:21 Bound to Virtual-Access1 (down, cloned from Virtual-Template200) cir 364800bc 3648 be 0 byte limit 456interval 10 mincir 364800byte increment 456 Adaptive Shaping none pkts 216 bytes 11256 pkts delayed 0 bytes delayed 0 shaping inactive traffic shaping drops 0 Queueing strategy: fifo Output queue 0/40, 0 drop, 0 dequeued HQ-RTR#sh HQ-RTR#show ppp mul HQ-RTR#show ppp multilink No active bundles No inactive multilink interfaces Q-RTR# HQ-RTR# HQ-RTR# HQ-RTR# HQ-RTR# HQ-RTR#sh run Building configuration... Current configuration : 4382 bytes ! ! Last configuration change at 21:17:23 UTC Thu Dec 17 2009 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ-RTR ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs-mz.124-20.T1.bin warm-reboot boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model memory-size iomem 20 network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice-card 0 no dspfarm dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! vtp mode transparent archive log config hidekeys ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 service mgcp ! controller T1 0/0/1 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! class-map match-any CONTROL match ip dscp cs3 match ip dscp af31 class-map match-any RTP match ip dscp ef ! ! policy-map POLICY-CHECK class RTP priority percent 33 compress header ip rtp class CONTROL bandwidth percent 5 class class-default fair-queue ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex full speed 100 ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay no fair-queue frame-relay traffic-shaping frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template200 class traffic-shape ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.1 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output POLICY-CHECK ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ! ! no ip http server ip http authentication local no ip http secure-server ! ! map-class frame-relay traffic-shape frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 ! ! ! ! ! ! control-plane ! ! ! voice-port 0/0/0:23 ! ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ccm-manager music-on-hold ! mgcp
[OSL | CCIE_Voice] HARDWARE VPN
Hello, I use cisco 1811 router for my vpn connection but keep getting this on the console and i can connect EZVPN: User connect request ignored,tunnel IPx-Voice-vRack endpoint not ready for request Any one with similar experience and how was it resolved thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA
Hello, I have the same experience with the hq pstn phone . if i use the other ones like br1 or br2 pstn phone, it does not prompt me for the remote destination number On Thu, Dec 17, 2009 at 12:07 AM, Vccie Vccie voiceccie2...@gmail.comwrote: OK, I have SNR working with MVA working but I just can’t figure out why the Remote Profile is not matched. I can manually enter the remote destination number in for authentication and am able to make phone calls but I don’t want the –Enter Remote destination prompt-. The PSTN phone is sending the exact number of the remote destination. (there is no 9 on the remote destination number for dial-plan matching, as that is done with outbound translation patterns) There is no hair pining going on, it’s a straight h.323 gateway with dial-peers to the UCM. With partial match of 10 on incoming Service Parameters, the inbound Gateway doesn’t have any Calling Xlate patterns or and globalization that happens. So It shouldn’t be getting lost in the UCM. I have read Features-Service/SRND/and a few more doc’s on MVA and everything I read said it should work by matching inbound CLID but for some reason it’s not working for me…. Ooh and I have restarted service/UCM/GW a few times already.. Just if anyone has any thoughts. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] EM Enabled phone causing mobile connect to fail
Hello, the Mobility works because you have attached the owner id to the phone and the feature is not available on the device profile used during extension mobility am bothered about similar issues when phones at hq logs in at branch 1, they cant make calls as they used to because the device pool they use is attached to the phone and not DN or device pool Wondering if there is a work around On Tue, Dec 15, 2009 at 4:46 PM, kill mill jha...@gmail.com wrote: hi, Did any one face this issue ? If i enable the EM on a phone then the mobility softkey for voice moblle connect states I am You are not a valid Mobile Phone User If i disable it it works. Is there something i am missing ? Thnx ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Presence
Hello, Working on the presence question. i want to achieve desktop control, i added the phone to be controlled under the CtiGw on the callmanager and enabled the enduser on the Presence for desktop contol. After doing this phone can not dial and you cant call the phones, same applies to the cupc Also the cupc does not show the domain name on it when i try to switch to desktop, it show conectioing on the Health server and goes to disable Anyone with an idea of what is wrong Note Before adding the desktop features above everything except the desktop worked well ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] presence client
Hello, After the integration of the presence server to the callmanager. using the cupc client as a desktop control for hq phone 2 i encountered the following issues 1. the CUPC does not show the domain name entered in the presence server on the phone. it just displays the extension 2. i modified the host file name on the PC that hat has the CUPC, still no luck tested by doing nslookup to the hostname, and it was resolving Anyone with a fix thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Looking for CUE license
Hello, i was able to download it. i went the the the UCCX Server and changed the username and password to cisco and cisco respectively i have been using ipexpert and cisco and documented in the PG thanks On Wed, Dec 9, 2009 at 5:29 PM, Omotayo adefilabi...@gmail.com wrote: the licenses are there and still got the errors i will try again today On Wed, Dec 9, 2009 at 3:09 PM, Bas Janssen basmj...@msn.com wrote: Hello, This might be because FTP is not running / installed on that server image. Do a RDP session to UCCX to see if the licenses are there. If not revert ONLY UCCX and load ILT lab 1 for ONLY UCCX. Then you can download the license. (I did this yesterday) regards, Bas -- Date: Wed, 9 Dec 2009 02:58:12 +0100 Subject: Re: [OSL | CCIE_Voice] Looking for CUE license From: adefilabi...@gmail.com To: o...@ipexpert.com CC: basmj...@msn.com; ccie_voice@onlinestudylist.com Hello, I tried but keep getting error messages CUE# $.10.210.5/cue-vm-license_12mbx_ccm_7.0.1.pkg username ipexpert ? password password on server cr CUE# $m-license_12mbx_ccm_7.0.1.pkg username ipexpert password cisco ? cr CUE# $m-license_12mbx_ccm_7.0.1.pkg username ipexpert password cisco WARNING:: This command will install the necessary software to WARNING:: complete a clean install. It is recommended that a backup be done WARNING:: before installing software. Would you like to continue? [n]y Downloading ftp cue-vm-license_12mbx_ccm_7.0.1.pkg Error: Download error Can not download cue-vm-license_12mbx_ccm_7.0.1.pkg error code 530 : error type 'Access denied: 530' On Tue, Dec 8, 2009 at 7:31 PM, Otto Sanchez o...@ipexpert.com wrote: Hello Bas, Try reverting the uccx server, if the file is still not there and ftp is not up and running, you may get the file from proctorlabs support, set your own ftp server and load the desired CUE license from there, HTH, On Tue, Dec 8, 2009 at 10:47 AM, Bas Janssen basmj...@msn.com wrote: Hi, I am working on lab 11a vol 1 and I am looking for the BR2 CUE license file. It is on not flash and I also cannot find it on 10.10.210.5 . FTP is not running and I cannot locate the file via the Remote Desktop. Has the procedure been changed? CUE is now in CCM mode Regards, Bas -- Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®.http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CCX AGENTS
Hello, I have subscribed a phone to the contact center agent service when i click on the service button on the phone, i get the message Host not found Any one witha suggestion thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCX AGENTS
it was the variance is port number. it works well. Though i configured i as a button login when i press the button, its still request for Ext Pwd and ID What do you think On Thu, Dec 10, 2009 at 1:50 AM, Tanner Ezell tanner.ez...@gmail.comwrote: You probably used a host name instead of an IP address and your phones are unable to resolve the name to an IP. On Wed, Dec 9, 2009 at 7:30 PM, Omotayo adefilabi...@gmail.com wrote: Hello, I have subscribed a phone to the contact center agent service when i click on the service button on the phone, i get the message Host not found Any one witha suggestion thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Tanner Ezell ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPexpert Web site - still issues?
it didnt work I feel they should not have launched the Site until its fully functional i cant access my documents with userid and email id i have mailed the support but no one is responding On Mon, Dec 7, 2009 at 8:15 PM, Steve Sarrick ssarr...@drsllc.net wrote: Here is the response I received and it worked fine for me IPexpert is currently in the process of transferring all client product files/information over to our new website. We are working hard to ensure that this process goes as quickly and smoothly as possible. Unfortunately, for the time being, this means that some of our clients may not have access to the files in their account and have issues logging into the new site until all transfers have been completed. We sincerely apologize for the inconvenience this may have caused you. However, there is a way to get access to the old IPexpert member’s site but you will have to jump through a couple of hoops. First you will need to disable on the browser you are using, the option to check server certificate revocation. In Internet Explorer go to Tools - Internet Options - Advanced (Tab) - Under “Settings” Scroll down to Security - Uncheck “Check for server certificate revocation* Then navigate back to the following link and bypass the security warning. In Fire Fox go to Tools - Options - Advanced - Encryption (Tab) - Validation (Button) - Uncheck “Use the Online Certificate Status Protocol…” Then navigate back to the following link and add an exception for our website and bypass the security warning. In Google Chrome go to Customize and Control Google Chrome (Wrench Icon) - Options - Under the Hood(Tab) - Scroll down page to Security - Uncheck “Check for server certificate revocation Then navigate back to the following link and bypass the security warning. URL: ipxweb001.ipexpert.com You will then be at a basic login page where you will be able to login to the old IPX members’ page and download your ebooks using your old IPX login credentials. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Steve Denney (stdenney) *Sent:* Monday, December 07, 2009 2:04 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] IPexpert Web site - still issues? Sorry to clog up the study list, but like (apparently) many others today, I’m having ongoing issues with the new IPexpert Web site. Response / refresh time is horrible, with pages refreshing unacceptably slow, or not correctly at all (missing images, etc.). Also, have not been able to log in to the Members area, using either my old user ID or my email address. Thinking that you might want to consider a backout / contingency plan at this point, folks...until the new site has been properly tested... Regards, Steve Denney, CISSP Systems Engineer - Technology Solutions Network Voice and Unified Communications Products Cisco Systems, Inc. 125 High Street, 21st Floor Boston, MA 02110 stden...@cisco.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPexpert Web site - still issues?
Hello, Thanks Why is Vol 2 Voice WB. an exception? On Tue, Dec 8, 2009 at 2:05 PM, Wayne Lawson groupst...@ipexpert.comwrote: Omotayo - (and others), Due to the structure and what's happening on the backend - there was no way to run 2 different systems concurrently. Everything will be available and back to normal (actually better than normal) today. We will also be adding additional labs / sections as almost every workbook is now completed with the exception of our Vol 2 Voice WB. Also (Omotayo) - I apologize for our support team not getting back to you. They have been very responsive - I'm sure your email slipped through the cracks (which is still unacceptable). Gang - I apologize for the major inconvenience, but appreciate the support and patience - we're really working on adding some great features and I feel that it will be worth the few days of downtime once this is up and functioning as designed. Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 101 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com. On Dec 8, 2009, at 6:49 AM, Omotayo adefilabi...@gmail.com wrote: it didnt work I feel they should not have launched the Site until its fully functional i cant access my documents with userid and email id i have mailed the support but no one is responding On Mon, Dec 7, 2009 at 8:15 PM, Steve Sarrick ssarr...@drsllc.net wrote: Here is the response I received and it worked fine for me IPexpert is currently in the process of transferring all client product files/information over to our new website. We are working hard to ensure that this process goes as quickly and smoothly as possible. Unfortunately, for the time being, this means that some of our clients may not have access to the files in their account and have issues logging into the new site until all transfers have been completed. We sincerely apologize for the inconvenience this may have caused you. However, there is a way to get access to the old IPexpert member’s site but you will have to jump through a couple of hoops. First you will need to disable on the browser you are using, the option to check server certificate revocation. In Internet Explorer go to Tools - Internet Options - Advanced (Tab) - Under “Settings” Scroll down to Security - Uncheck “Check for server certificate revocation* Then navigate back to the following link and bypass the security warning. In Fire Fox go to Tools - Options - Advanced - Encryption (Tab) - Validation (Button) - Uncheck “Use the Online Certificate Status Protocol…” Then navigate back to the following link and add an exception for our website and bypass the security warning. In Google Chrome go to Customize and Control Google Chrome (Wrench Icon) - Options - Under the Hood(Tab) - Scroll down page to Security - Uncheck “Check for server certificate revocation Then navigate back to the following link and bypass the security warning. URL: ipxweb001.ipexpert.com You will then be at a basic login page where you will be able to login to the old IPX members’ page and download your ebooks using your old IPX login credentials. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Steve Denney (stdenney) *Sent:* Monday, December 07, 2009 2:04 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] IPexpert Web site - still issues? Sorry to clog up the study list, but like (apparently) many others today, I’m having ongoing issues with the new IPexpert Web site. Response / refresh time is horrible, with pages refreshing unacceptably slow, or not correctly at all (missing images, etc.). Also, have not been able to log in to the Members area, using either my old user ID or my email address. Thinking that you might want to consider a backout / contingency plan at this point, folks...until the new site has been properly tested... Regards, Steve Denney, CISSP Systems Engineer - Technology Solutions Network Voice and Unified Communications Products Cisco Systems, Inc. 125 High Street, 21st Floor Boston, MA 02110 stden...@cisco.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training
[OSL | CCIE_Voice] issues with my ucxx lab
Hello, when i opened the CRS Editor, i can not drag icons from the left to the design pallete. it does not move across Aany one with an idea of whta the problem might be thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] issues with my ucxx lab
Hello Not to worry. its fine now Thanks On Mon, Dec 7, 2009 at 12:03 PM, Omotayo adefilabi...@gmail.com wrote: Hello, when i opened the CRS Editor, i can not drag icons from the left to the design pallete. it does not move across Aany one with an idea of whta the problem might be thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] new site looks great
I can not log in too On Sat, Dec 5, 2009 at 12:10 AM, Leslie Meade lme...@signal.ca wrote: I see the web site is up... Looks very swish Great work, but FYI I am unable to log into the site :) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
