[OSL | CCIE_Voice] regarding ntp server
We are installing a new local Cisco Call Manager server and would like to point it to a global NTP server so that the devices will be in sync. Regularly, one would use a windows DC or even PDC, but we were told that the Call Manager needs a Linux/Unix/Cisco device to serve as NTP server. Has any one came across NTP server issue. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] regarding ntp server
Thanks Juan. Just want to know when installing CUCM 8.x on VMware ,for NTP server do we need windows server only (NTPv4 server). On Wed, Jul 4, 2012 at 5:55 PM, Juan Carlos Anzola juancarlosanz...@gmail.com wrote: Hi Brajesh, There is no need to give access for CUCM to a public NTP Server. What i normally do in those cases is to sync an IOS Router or Switch (if supported) with the internal NTP Server, and then point CUCM to that IOS Device. That way you can still sync to your internal NTP in a two-step fashion, avoiding any issue that may arise when using Windows NTP Servers. HTH Juan Carlos On Wed, Jul 4, 2012 at 7:36 AM, brajesh kumaR brjku...@gmail.com wrote: We are installing a new local Cisco Call Manager server and would like to point it to a global NTP server so that the devices will be in sync. Regularly, one would use a windows DC or even PDC, but we were told that the Call Manager needs a Linux/Unix/Cisco device to serve as NTP server. Has any one came across NTP server issue. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Juan Carlos Anzola ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Intersite calls MOH
Thanks Vik. In my case moh from branch router started working for external users when ip multicasting enabled on switch.It seems when enabling multicasting on switch for both sites MOH started working between two location also. On Sun, Jan 22, 2012 at 12:54 PM, Vik Malhi vma...@ipexpert.com wrote: We set the VGW as a multicast server- so this works for IP phones and PSTN calls. You must ensure you specify the Voice SVI IP Address and Loopback address in the multicast moh route VlanXX LoXX statement within telephony-service/call-manager-fallback. Technically this is not multicast routing- there is no routing since the VGW is itself the MOH server. You can disable multicast-routing / PIM and it will still work. On Jan 18, 2012, at 8:34 AM, brajesh kumaR wrote: Hello, If branch site VGW configured for MOH multicast from branch router/VGW so will inter site IP calls ( two different location calls) will also get multicast from voice gateway or moh from multicast only works for external PSTN calls. Will on-net calls use VGW multicast MOH from flash?? During inter site calls I found following on VGW. Multicast IP showinng 0. Jan 18 14:40:52.381: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.381: moh_process_ccb: dstadr 0.0.0.0, callid 55765, port 0, codec 65535, moh_en 0, moh_addr 0.0.0.0 Jan 18 14:40:52.381: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.473: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.497: moh_process_ccb: dstadr 10.136.170.197, callid 55765, port 20650, User phone IP address : codec 5, moh_en 0, moh_addr 0.0.0.0 .Multicast address coming as 0. Jan 18 14:40:52.497: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52: %ISDN-6-CONNECT: Interface Serial0/0/0:27 is now connected to 0170480054 N/A Jan 18 14:40:52.533: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.537: moh_process_ccb: dstadr 10.136.170.197, callid 55765, port 20650, codec 5, moh_en 0, moh_addr 0.0.0.0 Jan 18 14:40:52.537: moh_update_rtp: callID 55764 dstCallID 55765 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Intersite calls MOH
Hello, If branch site VGW configured for MOH multicast from branch router/VGW so will inter site IP calls ( two different location calls) will also get multicast from voice gateway or moh from multicast only works for external PSTN calls. Will on-net calls use VGW multicast MOH from flash?? During inter site calls I found following on VGW. Multicast IP showinng 0. Jan 18 14:40:52.381: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.381: moh_process_ccb: dstadr 0.0.0.0, callid 55765, port 0, codec 65535, moh_en 0, moh_addr 0.0.0.0 Jan 18 14:40:52.381: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.473: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.497: moh_process_ccb: dstadr 10.136.170.197, callid 55765, port 20650, User phone IP address : codec 5, moh_en 0, moh_addr 0.0.0.0 .Multicast address coming as 0. Jan 18 14:40:52.497: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52: %ISDN-6-CONNECT: Interface Serial0/0/0:27 is now connected to 0170480054 N/A Jan 18 14:40:52.533: moh_update_rtp: callID 55764 dstCallID 55765 Jan 18 14:40:52.537: moh_process_ccb: dstadr 10.136.170.197, callid 55765, port 20650, codec 5, moh_en 0, moh_addr 0.0.0.0 Jan 18 14:40:52.537: moh_update_rtp: callID 55764 dstCallID 55765 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] h323 gateway CME in same Router
Actually bind command is under voice service voip/sip and source-address command is defined under voice register global and it seems only one of the command is enough but if we have defined both it should be the same IP. On Sat, Dec 31, 2011 at 8:28 PM, datucha123 datucha123 datucha...@gmail.com wrote: If you are not using any H323 trunks (like H323 trunk to ITSP or another UCME) then you do not need the H323 Binding and Interface at all. SCCP IP Phone are POTS Dial-peers, so when you make a call from SCCP IP Phone to TDM PSTN, you can think of this call like a TDM to TDM call, where you do not need any H323 or SIP interfaces. IPPhone --sccp signalling -- ip x.x.x.x --- analog connection - Telco Where IP x.x.x.x is the IP address of telepony-service. Almost the same goes the UCME SIP Phones, but instead of POTS dial-peers, the SIP Phone are using VoIP Dial-peer. At this point, you have to consider the SIP Bindings to an interface, where you will get the VoIP to TDM call. IPPhone --SIP signalling -- ip x.x.x.x --- analog connection - Telco Where IP x.x.x.x is the IP address of voice register global When using SCCP IP Phones, the RTP Stream is terminated at the Telephony-Service interface, before it goes out to PSTN. When using SIP IP Phones, the RTP Stream is terminated at the SIP interface (Bind command), before it goes out to PSTN. On Sat, Dec 31, 2011 at 6:05 PM, Ken Wyan kew...@gmail.com wrote: It is common to terminate PSTN line ( analog or ISDN) to the same CUCME router. I have seen many people saying it as an h323 gateway but we don't use any h323 commands in this router. I think following is the call flow. (signalling) IPPhone --sccp signalling -- ip x.x.x.x --- h323 signalling --- ip y.y.y.y --- analog connection - Telco ip x.x.x.x ip y.y.y.y are defined as follows telephony-service ip source-address x.x.x.x interface a /a ip address y.y.y.y 255.255.255.0 h323-gateway voip bind srcaddr y.y.y.y ( normally this command is not used if CME is in the same router ) ip pim dense-mode ( to send multicast moh to PSTN callers ) Please confirm whether the above signalling path is correct or otherwise please correct me. (above is purely my assumption) Where does rtp stream terminate ( which ip ) before converting to TDM or analog ? Is it loopback interface ? Can we change it? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Calls to CUE fail over WAN
But as per voice class codec still preference one is g711ulaw codec which is on incoming dial-peer same codec as outbound peer. On Wed, Dec 28, 2011 at 2:00 PM, datucha123 datucha123 datucha...@gmail.com wrote: Nice to hear that you have solved the issue. The solution is correct. When you a have a Voice-Class Codec that has the G711 in it, then the Xcoder is not invoked as the G711 is supported by the incoming dial-peer. In such cases, you have to force the incoming dial-peer from CUCM (or any other sources) to use the G729, so that xcoder will get invoked. On Wed, Dec 28, 2011 at 4:44 AM, John McGaughey (jomcgaug) jomcg...@cisco.com wrote: Nevermind. I figured it out. I created a separate inbound dial-peer with g729 and that caused the xcoder to get invoked. voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 dial-peer voice 36002 voip voice-class codec 1 incoming called-number 3600 dtmf-relay rtp-nte 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 Before I had it with only this dial-peer. dial-peer voice 3600 voip destination-pattern 3600 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 3600 dtmf-relay sip-notify codec g711ulaw no vad I guess if the inbound dial-peer can’t satisfy the codec it fails. But if it does, and the outbound dial-peer does not, then it invokes the xcoder. John From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] Sent: Tuesday, December 27, 2011 6:42 PM To: John McGaughey (jomcgaug); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Calls to CUE fail over WAN Did you enable cube? Sent from my Verizon Wireless Phone - Reply message - From: John McGaughey (jomcgaug) jomcg...@cisco.com Date: Tue, Dec 27, 2011 7:27 pm Subject: [OSL | CCIE_Voice] Calls to CUE fail over WAN To: ccie_voice@onlinestudylist.com Hello, I'm in Workbook 2, Lab 8, question 4.5. I have CUE setup properly and calls between CME phones roll successfully to voicemail. However, they fail to roll to voice mail when an HQ phone calls a BR2 phone. The answer guide says to configure an xcoder on the BR2 router. Here's my config. sccp local Vlan400 sccp ccm 10.10.202.1 identifier 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register br2-xcoder ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP telephony-service sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 br2-xcoder authentication credential admin cisco xml user pvadmin password cisco 15 max-ephones 10 max-dn 10 no-reg both ip source-address 10.10.202.1 port 2000 url services http://10.10.202.2/voiceview/common/login.do url authentication http://10.10.202.1/CCMCIP/authenticate.asp pvphone cisco voicemail 3600 max-conferences 12 gain -6 web admin system name admin password cisco transfer-system full-consult create cnf-files version-stamp 7960 Dec 28 2011 00:02:05 Here's the dial-peer to CUE. dial-peer voice 3600 voip destination-pattern 3600 session protocol sipv2 session target ipv4:10.10.202.2 incoming called-number 3600 dtmf-relay sip-notify codec g711ulaw no vad A sh sccp shows the xcoder registered. BR2-RTR#sh sccp SCCP Admin State: UP Gateway Local Interface: Vlan400 IPv4 Address: 10.10.202.1 Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 10.10.202.1, Port Number: 2000 Priority: N/A, Version: 7.0, Identifier: 1 Trustpoint: N/A Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.10.202.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 A trace on the router shows this. Dec 28 00:00:52.604: //-1//SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:3600@10.10.202.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.200.3:5060;branch=z9hG4bK2AB55 Remote-Party-ID: HQ Phone 2 sip:5002@10.10.200.3;party=calling;screen=yes;privacy=off From: HQ Phone 2 sip:5002@10.10.200.3;tag=4F3F96E8-171B To:
Re: [OSL | CCIE_Voice] ICT vs H225
It is not fair to say Datucha123 just putting question on this forum.He has well presented questions and come back with answer with lot new things which is not clear if you look at first sight. If Datucha123 continues to be on his way of such fast learning he will be very near to ccie. On Tue, Dec 20, 2011 at 3:25 AM, Amit Singh batraji...@yahoo.com wrote: Mike I agree Datucha123 loves spoon feeding. Regards Amit Sent from my iPad On 20/12/2011, at 2:28 AM, Mike mik...@msn.com wrote: I gotta ask…are you using this list as your own study guide? I’ve watched you send over 200 questions to this list that can be easily answered in the Cisco documentation… From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of datucha123 datucha123 Sent: Monday, December 19, 2011 5:39 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] ICT vs H225 Hello, Please, can you recomment which one must be used in which situation - ICT vs H225 Trunk. First of all, I know that the SRND talks about that, but that is more the theoretical one, whereas I need some real practical recommendation. So the first thing that I have tested for differences between ICT and H225 Trunks: All H323 connections are made between CUCM and CUCME, either directly or through gatekeeper or even CUBE. Direct connect to from CUCM to CUCME. ICT (Non-GK controlled) - Does not support Supplementary Services (like hold, transfer and etc) unless the MTP is checked. The CUCME Admin guide says to use this Trunk type to connect CUCM to CUCME and check the MTP on CUCM ICT Trunk. H323 Gateway - when this type of H323 connection is made to CUCME, the CUCM IP phone support Supplementary Service even without the MTP checked. Connect to CUCME through CUBE H323 Gateway - wait for far end H245 TCS Unchecked. When using this type of connetion, I have enabled the emptycapablility command on CUBE, so that CUCM IP Phones could support Supplementary Service without MTP. While when using MTP checked, it does not matter any more whether the emptycapablility is enabled or not, the Supp Service are operational on CUCM IP Phones. ICT (Non-GK controlled) - When using this type of connection to CUBE, CUCM IP Phones does not support Supplementary Services (even the emptycapablility is enabled on CUBE), until I check the MTP for this ICT. Connect to CUCME using Gatekeeper ICT (Gatekeeper-controlled) - I beleive, when using this type of connection, it has the same charasteristics as described already above for ICT Trunks. (Not Supp-service unless the MTP is used) H225 Trunk (Gatekeeper-controlled) - I beleive, when using this type of connection, it has the same charasteristics as described above for H323 Gateway Type. Based on that, can you please recommend, at least for the exam, which type of Connection has to be used for those kinds of H323 connectivity between CUCM and CUCME? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ICT vs H225
H323 GW will send RAS request to that ip which is defined using H323-gateway voip id command on GW. On GK when defining local zone IP address is not mandatory. If you have not defined any binding to h323 GW then it will probably take serial interface ip and send ras request to ip defined in H323-gateway voip id command. On Mon, Dec 19, 2011 at 4:09 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, Please, can you recomment which one must be used in which situation - ICT vs H225 Trunk. First of all, I know that the SRND talks about that, but that is more the theoretical one, whereas I need some real practical recommendation. So the first thing that I have tested for differences between ICT and H225 Trunks: All H323 connections are made between CUCM and CUCME, either directly or through gatekeeper or even CUBE. Direct connect to from CUCM to CUCME. ICT (Non-GK controlled) - Does not support Supplementary Services (like hold, transfer and etc) unless the MTP is checked. The CUCME Admin guide says to use this Trunk type to connect CUCM to CUCME and check the MTP on CUCM ICT Trunk. H323 Gateway - when this type of H323 connection is made to CUCME, the CUCM IP phone support Supplementary Service even without the MTP checked. Connect to CUCME through CUBE H323 Gateway - wait for far end H245 TCS Unchecked. When using this type of connetion, I have enabled the emptycapablility command on CUBE, so that CUCM IP Phones could support Supplementary Service without MTP. While when using MTP checked, it does not matter any more whether the emptycapablility is enabled or not, the Supp Service are operational on CUCM IP Phones. ICT (Non-GK controlled) - When using this type of connection to CUBE, CUCM IP Phones does not support Supplementary Services (even the emptycapablility is enabled on CUBE), until I check the MTP for this ICT. Connect to CUCME using Gatekeeper ICT (Gatekeeper-controlled) - I beleive, when using this type of connection, it has the same charasteristics as described already above for ICT Trunks. (Not Supp-service unless the MTP is used) H225 Trunk (Gatekeeper-controlled) - I beleive, when using this type of connection, it has the same charasteristics as described above for H323 Gateway Type. Based on that, can you please recommend, at least for the exam, which type of Connection has to be used for those kinds of H323 connectivity between CUCM and CUCME? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] GK Binding
H323 GW will send RAS request to that ip which is defined using H323-gateway voip id command on GW. On GK when defining local zone IP address is not mandatory. If you have not defined any binding to h323 GW then it will probably take serial interface ip and send ras request to ip defined in H323-gateway voip id command. On Tue, Dec 20, 2011 at 8:33 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, I have tested, and when configuring the Gakeeper, and specifying the IP address for the first Zone Local, the Router binds the RAS signaling from the Gatekeerer Router to this IP address. Although I have not configured the interface with h323-gateway voip interface or h323-gateway voip bind srcaddr. Please can you confirm that the IP Address in the Zone Local configuration is used to Bind the RAS signaling to that IP address that leaves the Gatekeeper Router. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] How to verify dtmf in voice gateway
Use debug voip ccapi inout and look for digit= to verify DTMF digits sent. You can debug this live on gateway and verify any DTMF digits entered in between. On Fri, Dec 9, 2011 at 12:38 PM, So Gwaai sogw...@gmail.com wrote: Anyone know how to verify the voice gateway send the dtmf through the PRI port? Any debug command or ccm trace we can get? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] [EMEA Cluster Gateway Dial Plan Assistance Munich Germany]
Hi Micheal, Start with international ,national and mobile number calls dial plan with following. You need to manipulate called number either in CUCM or at gateway. For ease you can use 0T at POTS dial peer so that international call will be send with 00 prefix and national/mobile call with 0 prefix which also might be expected from PSTN provider. 0.00!# International calls 0.00!International calls 0.0[2-9]!# Germany national calls 0.0[2-9]!Germany national calls 0.01[5-7]!# Germany Mobile Numbers 0.01[5-7]! There are many non geographical numbers in Germany which you need to take care of if there is any dial plan requirement. Use following links to more about Germany dial plans and specific city codes perfixes. http://en.wikipedia.org/wiki/Telephone_numbers_in_Germany http://www.howtocallabroad.com/results.php?callfrom=indiacallto=germany Regards, Brajesh. On Thu, Dec 1, 2011 at 7:23 PM, michael.se...@compucom.com wrote: Greetings – I am seeking input on developing a dial plan for a site that has been thrown my way in Munich Germany. I’m new to ISDN ERA and have been using NANP for years. Any input regarding developing a dial plan for Munich Germany, including sample configurations of CUCM and Gateways, would be greatly appreciated or if you can point me to resources ( in the right direction) would be immensely appreciated. Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Multicast MOH with Branch IP Phones
Can you please check if branch sites phones have MRGL assigned. What is the region setting between BR1 phone and moh server.If it is g729 then transcoder will be invoked to send spoofed moh stream to branch phone using flash moh file. Regards, Brajesh. On Mon, Nov 28, 2011 at 7:06 AM, Pradeep Kumar Sharma sonu.netwo...@gmail.com wrote: Hello Experts, I am working on Branch site Multicast MOH example and spoofing the moh from router flash memory. My MOH server inter-region setting is G711 for all branch sites. When placing a PSTN call on hold from a Branch phone, the Moh spoofing is working perfectly. When placing a internal branch to branch call on hold, moh is not working and i am getting a dead-air. At the same time:- - There is no debug output on the branch router. (debug ccm-manager music-on-hold all) - show perf query class Cisco MOH Device on call manager CLI showing MOHMulticastResourceActive = 1 - ccm-manager music-on-hold applied on the Branch router. - multicast moh 239.1.1.1 port 16384 route #fast ethernet IP # #Loop0 IP# applied on branch router. - Only 1 MOH server in MRGL with Multicast enabled. (max hop = 1) - Multicast routing is disabled on HQ and Branch Site for moh spoofing. Any thought ? -SONU- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Unity Connection user sync issue
Hi Ken, Is this user import you are trying in lab environment (no AD) or production environment? Does your CUCM user id AD intergrated.If yes then you can use unity connection bulk tool to import all users from cucm (only if primary extension is set on user id) and use then in creating new mailboxes and updating the mailboxes. Assuming the Primary Extension gets set in Call Manager…. ( 1 ) On Unity Connection, go to Import Users and import the users from Call Manager . ( 2 ) Now you need to export a list of the users you just imported from Call Manager to get the user id's in the same case as they were created in AD. The reason they are correct in Call Manager is that it is also sync'd with AD. When you imported from Call Manager, Connection created the mailboxes with the same id as Call Manager pulled from AD. To get the list, use Connection's BAT tool to Export Users with Mailboxes.All you need is alias coloum.Take that column copy it into the first and second columns of a new Excel worksheet and save as CSV. Alias,LdapCcmUserId (These will be the two headers of excel/csv). ( 3 )Last step is to use the Connection Bulk Administration Tool to Update Users with Mailboxes, using the csv file you just created. That will cause them to be synchronized with AD. Please try and see if this fix your issue. Regards, Brajesh. On Mon, Nov 21, 2011 at 2:51 PM, Ken Wyan kew...@gmail.com wrote: Hi, I am still stuck with user import issue from CUCM to Unity Connection. In CUCM , I created username HQPH1 ( this is user id Last name of user , then I associated this username to phone line from CUCM GUI : Device --- Phone - Directory Number Configuration -- bottom of this page I added this username from user HQPH1 configuration page of CUCM , I added above phone as a controlled device ) Import of this user to CUC is successful ,,, but then I can see user alias is shown as hqph1 in CUC users. Cisco Unity Connection change username from upper case to lowercase when importing to user alias. Next , when I try to sync imported user , this step fails with error message shown below. Sunday, November 13, 2011 2:26:04 AM EST, HQPH1, banner.literal.data.error[An object with the SMTP address (hqph1@cuc7-pub) already exists.] All these things happenned when I was using Proctorlabs online racks. It happenned in different rack numbers in different sessions. I tried to re-create this issue with my VMWare CUC server. All the time it worked fine without any issues in user import / sync process. In my CUC server user alias is shown correctly in uppercase as well. As many of you are using Proctorlabs , please share with me if you faced this any workaround. I spent lot of time in proctorlabs to solve this , but no success yet. Please help me to find a solution. ( I re-re-re checked all user import / sync steps I followed the correct documented procedure ) On Sun, Nov 13, 2011 at 8:06 AM, Ken Wyan kew...@gmail.com wrote: Hi, I have configured users in CUCM ( each user associated with a phone , primary extension selected user is again associated under device line configuration) From Unity connection , Importing users works fine , but syncing users unsuccessful with following error. Sunday, November 13, 2011 2:26:04 AM EST, HQPH1, banner.literal.data.error[An object with the SMTP address (hqph1@cuc7-pub) already exists.] Sunday, November 13, 2011 2:26:04 AM EST, HQPH2, banner.literal.data.error[An object with the SMTP address (hqph2@cuc7-pub) already exists.] Even if sync fails , I can see all users under users tab of Unity Connection Admin GUI. Tried delete them import again , but same problem. (I tested this with Proctorlabs) Can anyone help me to find the cause please. Ken ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voice Class Codec Question
Hi Kshitij, Thanks for the great explanation.I have one doubt regarding case1. In case 1 (call coming from PSTN to branch phone) when we have not defined any codec on voip peer then by default g729 codec will be selected and it seems G729 codec finally used to connect to phone irrespective of fact that branch gateway and branch Ip phone are in same region. Show voice call summ on brach router shows that call connected using g729 and MOH to PSTN caller will not work in this case when CUCM is multicasting MOH (no multicating from branch router) as g729 codec is not supported. Same MOH/multicasting works when voip dial-peer codec is forced G711. PSTN GW --G729 --CUCMg729- IP PSTN-- GW-G729---IP Phone The reason for g729 codec used seems to be fact that this codec is used by GW voip dial-peer as default (only option) and now CUCM will negotiate codec with IP phone using sccp keeping in view that one leg is g729 so final selection will be g729. If this is the case transcoder should be invoked to send call to Ip phone. Has someone tested this in lab scenario. Regards, Brajesh. On Mon, Nov 14, 2011 at 6:06 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Hi Sonu, So for a call on the Branch site, I can think of a few scenarios: 1. Call coming in for an IP Phone at the branch site from the PSTN. 2. Call being made from a phone at Branch Site 1 to another phone at Branch Site 1. 3. Call being made from a phone at Branch Site X to a phone at Branch Site 1. 4. Call coming in for an IP Phone at the branch site from the PSTN rolling over to Unity. 5. Call coming in via a GK to a phone at Branch Site 1 For 2. and 3. we don't need to worry about the dial peers. This is SCCP signalling within CUCM itself and the codec selected is going to be governed by the Region settings. I am assuming that we have the following regions created (all sites are CUCM sites - if there is a GK involved, there might be a CME site in which case one of the Regions will not be there): Reg-SiteA Reg-SiteB Reg-SiteC Reg-GK Reg-MOH Within the same region, the relationship is G.711. Inter-region relationships are G.729. The only exception to this rule is the MOH region which is G.711 throughout. For 5. the dial-peer with a session target of ras shouldn't have any codec defined on it. That would invoke G.729r8 on such calls. For 1 and 2 we have the dial peer set up as you have described. In such a case, the Destination phone will be in Reg-SiteB and the ingress GW will also be in the same Region. So it doesn't really matter how we specify the voice class codec since this is not a call between sites. For 4, Unity should be in Reg-SiteA and the IP Phone/Ingress GW in Reg-SiteB. Thus, even though G711ulaw will be advertised in the TCS to CUCM, only G.729 will be negotiated due to the Inter-region relationship defined. What we should be looking at are calls from Site A TEHO to a Site B PSTN phone (Or something similar). According to me, this is a call between sites and once again, we needn't worry about the preference of codecs in the voice class command since: Site A IP Phone (Reg-SiteA) will be calling the egress GW at Site B (Reg-SiteB). If the incoming dial peer at Site B has both codecs defined, the GW will send an H.245 TCS to CUCM advertising both codecs. However, CUCM will enforce the region relationship(s) mentioned previously and will thus negotiate only G.729. Note that the preference of the codecs in the voice class codec becomes a matter of concern only when something like this happens: 1. An H.323 endpoint advertises G.711 as first preference and then G.729 2. GW advertises G729 as first preference and then G.711. In this scenario, the MSD will be the tie breaker. In most cases, for a CUCM scenario, CUCM becomes the master and wins, so to speak although I don't know of any way to define a codec preference on CUCM as such. On Mon, Nov 14, 2011 at 12:18 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: [VM Ware Question] (Adam Thompson) 2. Voice Class Codec Question. (Pradeep Kumar Sharma) 3. Re: can not save script in script repository (=?gbk?B?YnJ1bm8=?=) 4. Re: Number of IP Phones in the lab (Google) 5. uccx Unified CM Telephony Subsystem gray out (=?gbk?B?YnJ1bm8=?=)
[OSL | CCIE_Voice] Getting CUPS/CUCM restart/reload reason
Hello , Is there any way to know using CLI for server restart reason for CUPS/CUCM. Regards, Brajesh. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST keepalive time
Hi Emauel, Please use following command to delay the fallback to CUCM Using the command will delay fallback to configured minute. ccm-manager switchback uptime-delay minutes Regards, brajesh. On Tue, Nov 8, 2011 at 10:14 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: Hello experts, I currently have a customer who has CUCMBE in his environment with SRST enabled. His voice gateway is over a WAN link, and the link is unstable. His phones keep registering back and forth, and now he wants to keep his phones in SRST mode for a little longer than usual. His configs are call-manager-fallback based (no CME as SRST)... How can I achieve this? Is it through CUCMBE or his CME? His gateway is MGCP. Thanks Antonio Emanuel Damasceno CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written) CompTIA Network+ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] h323 gw question
This is to match inbound call coming from CUCM. Dial-peer 3000 voip will be used to matched inbound voip calls as well as outbound voip calls. Inbound in terms to call coming from cucm to h323 GW and outbound in terms of call coming from PSTN/GW to CUCM for inbound phone calls. 2011/11/6 bruno bruno.juni...@gmail.com: hello guys, regarding the h323 gw , why we need incoming called-number . under dial-peer ? i saw vik create another dial-peer for match any inbound . dial-peer voice 3000 voip destination-pattern ^3...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.10.211 incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3001 voip preference 1 destination-pattern ^3...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.10.210 dtmf-relay h245-alphanumeric no vad -- Best Regards, Bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] h323 gw question
Matching default voip dial-peer 0 for inbound voip can also work fine but it has VAD enabled by default and can choose any codec which can cause issues. Specifying specific inbound dial-peer like 3000 will force no vad and choose codec defined in voice class as we are forcing to match these using this dialpeer. Regards, Brajesh. On Sun, Nov 6, 2011 at 1:27 PM, Shrini linuxbos...@gmail.com wrote: In my opinion it is not required. I never used unless I am using it for MVA. On 11/5/2011 10:00 PM, bruno wrote: hello guys, regarding the h323 gw , why we need incoming called-number . under dial-peer ? i saw vik create another dial-peer for match any inbound . dial-peer voice 3000 voip destination-pattern ^3...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.10.211 incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3001 voip preference 1 destination-pattern ^3...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.10.210 dtmf-relay h245-alphanumeric no vad -- Best Regards, Bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] transfer file to router with no tftp or ftp server
You can try to upload this file using RDP (sharing local drive) to any server which has tftp configured and is in network with router where you want to send file using copy tftp flash. Regards, Brajesh. On Sun, Nov 6, 2011 at 1:29 PM, Shrini linuxbos...@gmail.com wrote: Upload the file to CUCM tftp and on router copy tftp: flash: On 11/5/2011 3:11 AM, Michael Miller wrote: In the case that you need to put IOS on a switch that is at the bootloader prompt, you could use XMODEM and a console cable. The same goes for a router at rommon if you can't use tftp. On Sat, Nov 5, 2011 at 9:43 AM, rschukne...@gmx.de wrote: When you have a CUCM than you an TFTP-Server... /Robert Original-Nachricht Datum: Fri, 4 Nov 2011 10:57:33 -0700 (PDT) Von: John Smith cci...@yahoo.com An: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Betreff: [OSL | CCIE_Voice] transfer file to router with no tftp or ftp server Forgive my ignorance, but if you needed to transfer a file from a PC to a router and had no tftp server or ftp server on the PC, how could accomplish that? Thank you. -- Empfehlen Sie GMX DSL Ihren Freunden und Bekannten und wir belohnen Sie mit bis zu 50,- Euro! https://freundschaftswerbung.gmx.de ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUCM OS Account
Hi Saad, If you would like to create a new OS admin account and password, use the 'set account' CLI command. For example, in your case, you could enter 'set account newosadmin'. The output should appear as follows: admin:set account newosadmin Privilege Levels are: Ordinary - Level 0 Advanced - Level 1 Please enter the privilege level :1 Please enter the password :* re-enter to confirm :* Account successfully created Please try and confirm. Regards, Brajesh. On Tue, Nov 1, 2011 at 2:44 PM, Mohamed Saad e.mohamed.s...@gmail.com wrote: Hey Guys I want to create another Account to ( Admin ) for the OS of CUCM beside the default one which was created during initial setup ... is it possible and how ? also I want to create another admin account for web gui administration with full control I made a copy of the current admin user for the web gui and changed the name and the password for the new user under application user page but the thing is there is a tick box beside it which means it can be deleted ...! I want it be immune to deletion ... is it possible ? -- Thanks Best Regards The only way of finding the limits of the possible is by going beyond them into the impossible No matter how many goals you have achieved, you must set your sights on a higher one ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Vol 1 LAB 5.2 -- SoS
Hi Voiper, Since call is not hitting GW inbound voip peer ,CUCM is not able to make contact with GW through h323. You need to check if h323 GW is configured correctly in CUCM (check ip address in GW) and on gateway binded to source address either loopback or interface on gateway config on router. Regards, Brajesh. On Fri, Oct 28, 2011 at 4:44 AM, Voiper datapack...@gmail.com wrote: Any help most welcome Voiper On Thu, Oct 27, 2011 at 6:44 PM, Voiper datapack...@gmail.com wrote: Thanks John for the prompt reply and suggestion. * rg-hq was created for HQ rg-rl was created for BR1 * rl-local-gw standard local route group * RP 911 rl-local-gw * DP hq rg-hq br1 rg-br1 * inbound dial-p voice 100 with incoming number . configured dial-peer voice 1 pots incoming called-number . direct-inward-dial dial-peer voice 5000 voip destination-pattern 2123945... voice-class codec 1 session target ipv4:10.10.210.11 incoming called-number . dtmf-relay h245-alphanumeric no vad dial-peer voice 911 pots destination-pattern 911 port 0/0/0:23 forward-digits 3 * Serial0/0/0:23 unassigned YES NVRAM up up * HQ-RTR#sh contro t1 T1 0/0/0 is up. Applique type is Channelized T1 * Call Manager service rebooted (last effort) * debug isdn q931 and debug voip dial-peer shows nothing dial 911 from HQ-ph2 and BR1-ph2 nothing happening. I just don't know what am I missing? It is a pretty straight forward lab, infact the beginning of Call Routing! ! Voiper On Thu, Oct 27, 2011 at 2:29 PM, John Ciccone ccie.cicc...@gmail.com wrote: Voiper, Go back and verify your steps 1 and 2. 1) Created the RG, RL, RP as per guide 2) Added Local Route Group to Device Pools Generally speaking, when a lab question states that a call is to be routed out of the Local gateway that is a clue that they want you to use the Standard Local Route Group. In this case, you would create a route group (rg-hq) for the HQ router (10.10.200.3). This is the RG that is placed in the HQ Device Pool. You then create a Route List (rl-local) and select Standard Local Route Group. The 911 route pattern will use the rl-local in the Gateway/Route List selection box. Again, double check that you have all of the above correct. Another item to check is the CSS set on the HQ phone, but that's probably not the issue here, as you already stated that you have the 911 patern in the none partition. When you say that there is no debug, what debug commands are you refering to? Debug isdn q931 ? This will show call atempts out of the HQ router toward the PSTN. Also do a debug voip dialpeer as this will verify if the call is making its way into the router and what dial-peers it's attempting to use. Make sure you have an inbound dial-peer configured and are not relying on dial-peer 0. On Thu, Oct 27, 2011 at 11:33 AM, Voiper datapack...@gmail.com wrote: Greetings to all: I seek help from those who have tread the path. Workbook Volume 1, lab 5.2 and have am already stuck :( Followed the PG and the walk through video with little success. Question 5.2 - All calls from UCM phones to Emergency Services must be routed out of the Local gateway. - Emergency Services can be dialed by entering 9-1-1 - The ANI should be in full E168 format - the +, country code and the national digits should be sent to the PSTN - You should configure the phones such that the telephone number in the top right of the screen of the phone displays the full E164 number 1) Created the RG, RL, RP as per guide 2) Added Local Route Group to Device Pools 3) voice translation-rule 911 rule 1 /^1/ /+1/ ! voice translation-profile ANI-OUT translate calling 911 4) dial-peer voice 911 pots translation-profile outgoing ANI-OUT destination-pattern 911 port 0/0/0:23 forward-digits all 5) HQ-RTR#sh isdn stat Global ISDN Switchtype = primary-ni ISDN Serial0/0/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) 6) Using None partition. (Also tried with pt-nanp-911) = When I dial 911 on HQ Phone 2, I get like busy tone. Apparently, the call doesn't reach the Gateway, and so there is no debug result. = The Gateways and the phone configurations are fine. No partitions used in phones. Routers have been reloaded too. = When i remove all the above and simply put the dial-peer 911, I can make 911 calls from any phone to the PSTN. I can also call from the PSTN. Any suggestion or tips would be thankful. Voiper ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
Re: [OSL | CCIE_Voice] B-ACD
Hi Kshitj, Is there any single pdf to get the link of EM, IPMA, IPPA ,IPPM in cisco.com using Products - - - Voice and Unified Communications ? Regards, Brajesh. On Tue, Oct 18, 2011 at 1:25 AM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Hi Ray, For configuring B-ACD, you can browse to: PSA webpage (this is available when you open IE. It may also be in the bookmark but looks a lot like the download pages. Remember that search is disabled on this, as are a few other options that are not specific to the exam): http://www.cisco.com/cisco/web/psa/configure.html Products - - - Voice and Unified Communications - - - - IP Telephony - - - - Call Control - - - Cisco Unified Communications Manager Express - - - Configuration Guides - - - - Cisco Unified CME B-ACD and Tcl Call-Handling Applications. In the page that opens, you need to go to:Cisco Unified CME Basic Automatic Call Distribution and Auto-Attendant Service (this is the same one that you have referenced) On Mon, Oct 17, 2011 at 9:30 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: SIP Traffic Classification On the Catalyst 3750 (Chris Martin) 2. Re: SRST Behaviour (Ashraf Ayyash) 3. B-ACD (Ray) -- Message: 1 Date: Mon, 17 Oct 2011 08:11:06 -0500 From: Chris Martin clm.c...@gmail.com To: AJ BG ciscoie2...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP Traffic Classification On the Catalyst 3750 Message-ID: CAHGbyRg8t7F+fY-vmqXUz=ndnzscb+szeuj-a0c7daakhmp...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 On 3750's if you cannot trust cos or dscp then to classify/mark traffic you need to use MQC like you have listed. If you want to classify SIP traffic in addition to SCCP, then yes 5060 would be required. Otherwise SCCP alone would be 2000. (2001 = SCCP analog gateways, 2002 = SCCP digital gateways, I doubt you will run into those in the lab) HTH, Chris On Sun, Oct 16, 2011 at 5:45 PM, AJ BG ciscoie2...@gmail.com wrote: Experts, On the Catalyst 3750, when there are both SCCP and SIP endpoints, what is the best way to classify control traffics without trusting the current marking? ? here is how I would do it : * ip access-list extended VVLAN-CALL-SIGNALING* * permit tcp any range 2000 2002 any * * permit tcp any any range 2000 2002* * permit udp any any eq 5060* * permit udp any eq 5060 any * ? Bellow is the config from IPX LAB 1 DSG: * access-list 102 permit tcp any any eq 2000* * access-list 102 permit tcp any eq 2000 any * Is it require to add UDP 5060? thanks, Joe ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20111017/267ed367/attachment-0001.html -- Message: 2 Date: Mon, 17 Oct 2011 09:48:54 -0500 From: Ashraf Ayyash ash.ayy...@gmail.com To: mgscip gpsvoiceexpe...@yahoo.com Cc: ccie ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST Behaviour Message-ID: CAEW==ns+mcOeYdVP8DVGrXOk=-+oztw902acnoxkvnp5ego...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 This can happen if you don't have enough DSP's in the router , can you try to do SRST in max dn 1 and max phone 1 ? Btw the fw of the phone cannot be the issue , the Ios us the had boy here :) Ash On Monday, October 17, 2011, mgscip gpsvoiceexpe...@yahoo.com wrote: Hi , I have some issue in SRST . When the Phones are get into SRST fallback-mode Phones didn't get any DN. I given the SRST mode auto-provision all , but i couldn't see any Ephone configuration in the running configuration. tried with Firmware upgrade , Reload the router but no luck. Thanks, Sriram.P -- next part -- An HTML attachment was scrubbed... URL:
