[OSL | CCIE_Voice] lab5.3 calling party number not work on sip phones, alway shows from E.164 number
Hi, When I tried lab 5.3, it works on SCCP phones, but not on SIP phones, though they use the same device pool with calling party transformation. On SIP phones, the from xxx where xxx is always E.164 number while on SCCP phone it depends on the PSTN location. Does any one have the same issue? I tried on latest SIP phone 9-2-3 on CCO, still the same. thanks, guoming ___ 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] BACD - question
so, its not really clear to me, what you would like to do/changeThe Thank you for calling is a audio file, which is made out of 2 config lines:param welcome-prompt _bacd_welcome.auparamspace english language enOut of this the file en_welcome-prompt_bacd_welcome.au (which is located in the configured directory flash:/bacdprompts) is played. If you want to change that you can either change the file itself or create a new one and refer to it. Take care of the en for the language at the beginning.By the way, is there a reason why you are using the flash BACD? As per my knowledge beginning from some IOS there is a build in application, which dont have to be refered to the script file in flash. Because it might happen to you in lab that the script (tcl) file doesnt exist.You can find the guide for this on the same place like the flash example. Its only little bit below the flash example. In the Bootcamp with Vik we got told from him to use the IOS one and not the Flash one.Martin Gesendet:Montag, 09. Januar 2012 um 23:07 Uhr Von:Randall Crumm rrcr...@yahoo.com An:Randall Crumm rrcr...@yahoo.com, Online Study ccie_voice@onlinestudylist.com Betreff:Re: [OSL | CCIE_Voice] BACD - question Maybe another way to ask is it looks like it is hitting the_bacd_options_menu.au which is already recorded.So I guess we would have to have the cli config to make the options menu if asked?Any thoughts?Randall From: Randall Crumm rrcr...@yahoo.com To: Online Study ccie_voice@onlinestudylist.com Sent: Monday, January 9, 2012 1:13 PM Subject: [OSL | CCIE_Voice] BACD - question HI,I have configured BACD on my sc-rtr. It does work but, I do not know how to control the greetings.When I dial 02077353000 i get Thank you for calling. This is excpected.Then I am gettingfor sales press 1for Customer Service press 2to dial by extension press 3for operator press 0Here is my config:applicationservice queue flash:/bacdprompts/app-b-acd-2.1.2.2.tcl param number-of-hunt-grps 2 param aa-hunt2 3002 param aa-hunt3 5010 param queue-len 15 param queue-manager-debugs 1!service aa flash:/bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 paramspace english language en paramspace english location flash:/bacdprompts/ param service-name queue param handoff-string aa param aa-pilot 3000 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 3600!dial-peer voice 222 voipservice aadestination-pattern 3000session target ipv4:10.10.110.3incoming called-number 3000dtmf-relay h245-alphanumericcodec g711ulawno vadThanks,Randall___For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comAre 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] CTI Route Point in Hunt Group
Hi all Does anyone have any idea why you can`t add a CTI RP to a Hunt Group ,when I add the CTI RP the Hunt Pilot keeps giving engaged tone ,but if I add a normal extension number it works fine ? Regards Rynard ___ 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] CTI Route Point in Hunt Group
HiRynard, You are not supposed to do that. CTI route points may not be associated with directory numbers (DNs) that are members of line groups and, by extension, that are members of hunt lists. If a DN is a member of a line group or hunt list, that DN cannot be associated with a CTI route point that you configure with the CTI Route Point Configuration window. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_1_1/ccmcfg/b06ctirp.html Cheers, Boris On Tue, Jan 10, 2012 at 8:21 PM, Rynard Coetzee rynard.coet...@bytes.co.zawrote: Hi all Does anyone have any idea why you can`t add a CTI RP to a Hunt Group ,when I add the CTI RP the Hunt Pilot keeps giving engaged tone ,but if I add a normal extension number it works fine ? Regards Rynard ** ** ___ 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] lab5.3 calling party number not work on sip phones, alway shows from E.164 number
I do not have such problems in my own lab. SIP phones Transform the ANI always. On Tue, Jan 10, 2012 at 12:19 PM, Guoming Zhang guozhang20...@yahoo.comwrote: Hi, When I tried lab 5.3, it works on SCCP phones, but not on SIP phones, though they use the same device pool with calling party transformation. On SIP phones, the from xxx where xxx is always E.164 number while on SCCP phone it depends on the PSTN location. Does any one have the same issue? I tried on latest SIP phone 9-2-3 on CCO, still the same. thanks, guoming ___ 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 http://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] BACD - question
Hi Randall, You cannot control which file is used for the options:it is hardcoded in the TCL script and not exposed as a param for us to change; i.e., there is no way to point the configuration to use another options audio file. To change the options menu you have to replace _bacd_options_menu.au file. If you have multiple AAs, then Cisco recommends to record all the greetings and menu choices into the welcome file (one file per AA that you configure) and record 2 seconds of silence for _bacd_options_menu.au. Anthony On Tue, Jan 10, 2012 at 5:13 AM, Randall Crumm rrcr...@yahoo.com wrote: HI, I have configured BACD on my sc-rtr. It does work but, I do not know how to control the greetings. When I dial 02077353000 i get Thank you for calling. This is excpected. Then I am getting for sales press 1 for Customer Service press 2 to dial by extension press 3 for operator press 0 Here is my config: application service queue flash:/bacdprompts/app-b-acd-2.1.2.2.tcl param number-of-hunt-grps 2 param aa-hunt2 3002 param aa-hunt3 5010 param queue-len 15 param queue-manager-debugs 1 ! service aa flash:/bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 paramspace english language en paramspace english location flash:/bacdprompts/ param service-name queue param handoff-string aa param aa-pilot 3000 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 3600 ! dial-peer voice 222 voip service aa destination-pattern 3000 session target ipv4:10.10.110.3 incoming called-number 3000 dtmf-relay h245-alphanumeric codec g711ulaw no vad Thanks, Randall ___ 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] Dummy CTI route points in Solutions Guide to CFA to Voice-mail
Hi, the solution guide uses dummy (unregistered) CTI route points in many tasks purely to forward calls (CFA) to Unity Connection, either mailboxes, live record, call handlers, greetings administrator etc Examples: Lab 1: Dummy CTI route point at DN 1113 for MeetMe task Why not just use a directory number (no device) to CFA to Voicemail? Is there any difference in using a directory number (no device) or a dummy CTI Route point? Anthony ___ 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] Dummy CTI route points in Solutions Guide to CFA to Voice-mail
In older version of CUCM, the DN could not be created without associating it to some Entity (Like IP Phone). In CUCM 7.0 the DN could be created withouth association to any entity, just standalone DN. In older versions of CUCM, if we wanted a Dummy DN, we had to associate it with some entity - CTI Route Point was a great way. In version 7.0, sure you can create a Dummy DN and configure the Forwards as necessary, no problem. Just the CTI Route Point version is still configurable, like in old days. On Tue, Jan 10, 2012 at 4:48 PM, Anthony Alba ascanio.al...@gmail.comwrote: Hi, the solution guide uses dummy (unregistered) CTI route points in many tasks purely to forward calls (CFA) to Unity Connection, either mailboxes, live record, call handlers, greetings administrator etc Examples: Lab 1: Dummy CTI route point at DN 1113 for MeetMe task Why not just use a directory number (no device) to CFA to Voicemail? Is there any difference in using a directory number (no device) or a dummy CTI Route point? Anthony ___ 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 http://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] BACD - question
Thaks Anthony for the confirmation Randall Sent from my iPhone On Jan 10, 2012, at 4:37 AM, Anthony Alba ascanio.al...@gmail.com wrote: Hi Randall, You cannot control which file is used for the options:it is hardcoded in the TCL script and not exposed as a param for us to change; i.e., there is no way to point the configuration to use another options audio file. To change the options menu you have to replace _bacd_options_menu.au file. If you have multiple AAs, then Cisco recommends to record all the greetings and menu choices into the welcome file (one file per AA that you configure) and record 2 seconds of silence for _bacd_options_menu.au. Anthony On Tue, Jan 10, 2012 at 5:13 AM, Randall Crumm rrcr...@yahoo.com wrote: HI, I have configured BACD on my sc-rtr. It does work but, I do not know how to control the greetings. When I dial 02077353000 i get Thank you for calling. This is excpected. Then I am getting for sales press 1 for Customer Service press 2 to dial by extension press 3 for operator press 0 Here is my config: application service queue flash:/bacdprompts/app-b-acd-2.1.2.2.tcl param number-of-hunt-grps 2 param aa-hunt2 3002 param aa-hunt3 5010 param queue-len 15 param queue-manager-debugs 1 ! service aa flash:/bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 paramspace english language en paramspace english location flash:/bacdprompts/ param service-name queue param handoff-string aa param aa-pilot 3000 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 3600 ! dial-peer voice 222 voip service aa destination-pattern 3000 session target ipv4:10.10.110.3 incoming called-number 3000 dtmf-relay h245-alphanumeric codec g711ulaw no vad Thanks, Randall ___ 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] Gatekeepers
Hi, I need to brush up my gwgk skills, I am really bad at it. Anyone would recommend books , videos , a particular lab? Thank you, -- Jeferson Guardia CCIE #28157 ___ 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] CME Multicast Paging: does it work to non-connected subnets?
I am configuring multicast paging on CME. ephone-dn 8 number no-reg primary name Sales Page paging ip 239.3.10.1 port 2000 ephone XX paging-dn 8 Two directly connected phones Ph1 Ph2 receive multicast paging and the RTP stream shows to 239.3.10.1. However, two phones Ph5 Ph6, not directly connected to CME show unicast streams. I have connected these two CME phones of HQ-RTR and configure HQ-RTR as multicast router. The multicast path to Ph5 Ph6 is working for multicast MOH For the paging multicast route I do not see any attempt by the phones to join 239.3.10.1. There is also no mroute on HQ-RTR. Anthony ___ 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] Gatekeepers
INE Deep Dives, and Advanced Technology Class videos. They are great, Also please refer to INE Blog for Gatekeepers: http://blog.ine.com/2009/01/15/gatekeeper-call-routing-at-a-glance/ On Tue, Jan 10, 2012 at 7:12 PM, Jeferson Guardia jefers...@gmail.comwrote: Hi, I need to brush up my gwgk skills, I am really bad at it. Anyone would recommend books , videos , a particular lab? Thank you, -- Jeferson Guardia CCIE #28157 ___ 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 http://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] CME Multicast Paging: does it work to non-connected subnets?
Remote CME IP Phone does not support Multicast. On Tue, Jan 10, 2012 at 7:26 PM, Anthony Alba ascanio.al...@gmail.comwrote: I am configuring multicast paging on CME. ephone-dn 8 number no-reg primary name Sales Page paging ip 239.3.10.1 port 2000 ephone XX paging-dn 8 Two directly connected phones Ph1 Ph2 receive multicast paging and the RTP stream shows to 239.3.10.1. However, two phones Ph5 Ph6, not directly connected to CME show unicast streams. I have connected these two CME phones of HQ-RTR and configure HQ-RTR as multicast router. The multicast path to Ph5 Ph6 is working for multicast MOH For the paging multicast route I do not see any attempt by the phones to join 239.3.10.1. There is also no mroute on HQ-RTR. Anthony ___ 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 http://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] MVA
I cannot understand the MVA configuration fully. I have read the Feature and Service Guide but could not understand. So here is the problem: 1) Do I have to configure the Mobile Voice Access Directory Number in the Media Resources the same as the H323 dial-peer has as the incoming called number? I have made some test, and it can be different then the Dial-peer Incoming Called Number and it works. Also when they both match, the MVA also works. I cannot get the idea which one must be set. 2) Do we need to enable the Enable Mobile Voice Access Service Parameter? Also I have tested and the MVA works always, whether that Service Parameter is True or False 3) What number must be written into Mobile Voice Access Number field in Service Paramters? And do we need it at all? I have tested it and the MVA works even when that fields is empty. ___ 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] SNR with SIP Trunk on CUBE SP
Hi guys, We have an issue of SNR not able to complete the calls to remote destination since the SIP Provider is authenticating on DIDs. The customer's CUCM is connected to our CUBE SP (ASR 1000) and we're connected to our SIP Provider. Example: I call into the phone 4242862304, it has SNR to 13102832382, but call fails to the Remote Destination because in the SIP message it shows my number as the calling party. CUCM 8.6.1 CUBE SP IOS 15.x SIP Provider We have tried setting these options in the SIP Trunk settings of CUCM. Outbound Calls Last Redirect Number (External) check Redirecting Diversion Header Deliver - Outbound I have a TAC case open but they're more confused than I am. I have also attached the debug. Maybe I'm missing some commands on the CUBE SP to allow Diversion Headers? Not sure, I'm new to CUBE SP. -- duy CCIE #27737 Voice INVITE sip:4242862304@10.127.159.241:5060 SIP/2.0Via: SIP/2.0/UDP 10.168.177.69:5060;branch=z9hG4bK5dgjsl300o0guk0dr0j0.1From: NGUYEN,DUYsip:8177143615@10.168.176.135;user=phone;tag=1268067308-1326161889877-To: En Pointe Technologiessip:4242862...@bluip.ecs.enpointe.comCall-ID: BW021809877100112-50676383@10.168.176.135CSeq: 582239531 INVITEContact: sip:8177143615@10.168.177.69:5060;transpor Jan 10 02:18:09.868: %SBC-5-MSG-5406-0605-80242F-1485: The Called-Party-ID has been obtained from the Request-URI. Called-Party-ID: 4242862304 Jan 10 02:18:09.869: %SBC-7-MSG-5001-0740-0B3AC0-3114: A header was not added to the SGM MAL during processing of an inbound SIP message. Header: Content-Disposition Jan 10 02:18:09.869: %SBC-5-MSG-5001-0108-56D003-0960: INVITE Jan 10 02:18:09.870: %SBC-7-MSG-3801-0044-4C8E7E-1737: SIP message sent to 0003 C7A8B145 (port 5060): SIP/2.0 100 TryingCall-ID: BW021809877100112-50676383@10.168.176.135CSeq: 582239531 INVITEFrom: NGUYEN,DUYsip:8177143615@10.168.176.135;user=phone;tag=1268067308-1326161889877-To: En Pointe Technologies sip:4242862...@bluip.ecs.enpointe.com;tag=sip+1+141d000e+2078b0d9Via: SIP/2.0/UDP 10.168.177.69:5060;received=10.168.177.69;branch=z9hG4bK5dgjsl300o0guk0dr0j0.1Server: CISCO-SBC/2.xContent-Length: 0 Jan 10 02:18:09.870: %SBC-5-MSG-1502-0098-7596A3-1360: Interworking Call Control has received a call setup message from a signalling stack Jan 10 02:18:09.870: %SBC-5-MSG-1502-0107-88D47D-1612: Interworking Call Control has sent a call setup ack message to a signalling stack Jan 10 02:18:09.871: %SBC-5-MSG-1502-0155-326D74-1440: SUBDB bypassed Jan 10 02:18:09.871: %SBC-7-MSG-1501-0306-C7EC65-1207: A config set was selected for use in RPS, based upon the configured admin domains (or default config set). If the chosen admin config set is zero, the default config set index is used instead. Selected config set index= 0 Actual config set index = 1 Selected priority= 0XC000 Selected admin domain (NULL if no AD MIB row was chosen) = BLUIP-Incoming Current policy type Jan 10 02:18:09.871: %SBC-7-MSG-1501-0107-E8D8A3-0122: Inbound number analysis begins. SBC Index = 0X0001 Config set Index = 0X0001 Input Called Address Type = 0X0003 Input Called Address = 4242862304 Input Calling Address Type = 0X0003 Input Calling Address = 8177143615 Jan 10 02:18:09.871: %SBC-7-MSG-1501-0108-E8D8A3-0626: Inbound number analysis succeeds. SBC Index = 0X0001 Config set Index = 0X0001 Last analysis table index = Last analysis table comment= Input Source Account = Input Source Adjacency = Input Source Address Type = 0X0003 Input Source Address = 8177143615 Input Called Address Type = 0X0003 Output Called Address Type = 0X0003 Input Called Address = 4242862304 Output Called Address Jan 10 02:18:09.872: %SBC-7-MSG-1501-0306-C7EC65-1207: A config set was selected for use in RPS, based upon the configured admin domains (or default config set). If the chosen admin config set is zero, the default config set index is used instead. Selected config set index= 1 Actual config set index = 1 Selected priority= 0X000A Selected admin domain (NULL if no AD MIB row was chosen) = BLUIP-Incoming Current policy type Jan 10 02:18:09.872: %SBC-7-MSG-1501-0111-7B8E34-0671: Routing begins. SBC Index = 0X0001 Config set Index = 0X0001 Calling Address Type = 0X0003 Called Address Type= 0X0003 Calling Address= 8177143615 Called Address = 4242862304 Jan 10 02:18:09.872: %SBC-7-MSG-1501-0112-7B8E34-0847: Routing succeeds. SBC Index =
Re: [OSL | CCIE_Voice] [cisco-voip] SNR with SIP Trunk on CUBE SP
John, I'll ask if that is an option. Joel, We're using a CUBE SP and there's no dial-peers. Checking on how to configure this thing on Cisco.com takes me no where. On Tue, Jan 10, 2012 at 8:44 AM, Joel Perez tman...@gmail.com wrote: Hi Duy, I've ran into similar issues and usually and i have solved it in my cases with the following: voice class sip-profiles 100 request INVITE sip-header Remote-Party-ID remove request INVITE sip-header Remote-Party-ID remove request INVITE sip-header Diversion add Diversion: sip:authorized DID@ipof outbound interface of cube Dial-peer voice 1 voip description to sip provider voice-class sip profiles 100 Hope this helps, Joel P On Tue, Jan 10, 2012 at 11:26 AM, ccieid1ot ccieid...@gmail.com wrote: Hi guys, We have an issue of SNR not able to complete the calls to remote destination since the SIP Provider is authenticating on DIDs. The customer's CUCM is connected to our CUBE SP (ASR 1000) and we're connected to our SIP Provider. Example: I call into the phone 4242862304, it has SNR to 13102832382, but call fails to the Remote Destination because in the SIP message it shows my number as the calling party. CUCM 8.6.1 CUBE SP IOS 15.x SIP Provider We have tried setting these options in the SIP Trunk settings of CUCM. Outbound Calls Last Redirect Number (External) check Redirecting Diversion Header Deliver - Outbound I have a TAC case open but they're more confused than I am. I have also attached the debug. Maybe I'm missing some commands on the CUBE SP to allow Diversion Headers? Not sure, I'm new to CUBE SP. -- duy CCIE #27737 Voice ___ cisco-voip mailing list cisco-v...@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip -- duy CCIE #27737 Voice ___ 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] SNR with SIP Trunk on CUBE SP
First of all, if you have an issue with Redirecting Number, then you can use the SIP Profile to correct the Diversion Header. And if you have an issue with ANI, then you have to set the Calling Party Transformation CSS for Remote Destination Profile. The Remote Destination Profile ignores any Calling Party Translation at Translation Pattern, Route Pattern, Route List, and even the Gateway/Trunk Calling Party Transformation and uses the Internal DN Extension. The only way to deal with Calling Party Translation when the call goes to Remote Destination Number is to use the Calling Party Transformation CSS at the Remote Destination Profile. Hope that helps. On Tue, Jan 10, 2012 at 8:26 PM, ccieid1ot ccieid...@gmail.com wrote: Hi guys, We have an issue of SNR not able to complete the calls to remote destination since the SIP Provider is authenticating on DIDs. The customer's CUCM is connected to our CUBE SP (ASR 1000) and we're connected to our SIP Provider. Example: I call into the phone 4242862304, it has SNR to 13102832382, but call fails to the Remote Destination because in the SIP message it shows my number as the calling party. CUCM 8.6.1 CUBE SP IOS 15.x SIP Provider We have tried setting these options in the SIP Trunk settings of CUCM. Outbound Calls Last Redirect Number (External) check Redirecting Diversion Header Deliver - Outbound I have a TAC case open but they're more confused than I am. I have also attached the debug. Maybe I'm missing some commands on the CUBE SP to allow Diversion Headers? Not sure, I'm new to CUBE SP. -- duy CCIE #27737 Voice ___ 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 http://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] SNR with SIP Trunk on CUBE SP
First of all, if you have an issue with Redirecting Number, then you can use the SIP Profile to correct the Diversion Header. And if you have an issue with ANI, then you have to set the Calling Party Transformation CSS for Remote Destination Profile. The Remote Destination Profile ignores any Calling Party Translation at Translation Pattern, Route Pattern, Route List, and even the Gateway/Trunk Calling Party Transformation and uses the Internal DN Extension. The only way to deal with Calling Party Translation when the call goes to Remote Destination Number is to use the Calling Party Transformation CSS at the Remote Destination Profile. Hope that helps. On Tue, Jan 10, 2012 at 8:26 PM, ccieid1ot ccieid...@gmail.com wrote: Hi guys, We have an issue of SNR not able to complete the calls to remote destination since the SIP Provider is authenticating on DIDs. The customer's CUCM is connected to our CUBE SP (ASR 1000) and we're connected to our SIP Provider. Example: I call into the phone 4242862304, it has SNR to 13102832382, but call fails to the Remote Destination because in the SIP message it shows my number as the calling party. CUCM 8.6.1 CUBE SP IOS 15.x SIP Provider We have tried setting these options in the SIP Trunk settings of CUCM. Outbound Calls Last Redirect Number (External) check Redirecting Diversion Header Deliver - Outbound I have a TAC case open but they're more confused than I am. I have also attached the debug. Maybe I'm missing some commands on the CUBE SP to allow Diversion Headers? Not sure, I'm new to CUBE SP. -- duy CCIE #27737 Voice ___ 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 http://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] Gatekeeper Codec control
Hello everyone. My question today is concerning controlling which codecs are used when utilizing RAS signaling via the Gatekeeper. I understand that I can control my codec inbound (at the BR2 CME) site via a inbound dial-peer that only utilizes g729r8. I also understand that using Transcoding at the same location will allow me to talk to a locally attached SIP phone (at CME site) that is configured to use g711ulaw only. However I am unclear as to how to program the CUCM controlled devices what codec to use when sourcing calls to the BR2 site via gatekeeper. If I am sourcing calls from CUCM across a gatekeeper trunk that has been configured to be in the HQ device pool which is associated with the HQ region which uses g711 intra-cluster... then shouldn't I be sourcing packets from CUCM to BR2 as g711ulaw? Any additional thoughts or clarification would be greatly appreciated. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 [cid:image001.gif@01CCCFA6.E516CFE0][cid:image002.gif@01CCCFA6.E516CFE0][cid:image003.gif@01CCCFA6.E516CFE0] Please note new e-mail address justin.mcint...@blackbox.commailto:alex.heve...@blackbox.com This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. inline: image001.gifinline: image002.gifinline: image003.gif___ 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] Gatekeeper Codec control
There is not way to force Codec in CUCM (except setting the lowest bandwidth consumption codec) When you configure Regions settings in CUCM, that actually sets the Maximum allowed codec within/between regions. Not the Codec itself. So for instance, when you set G711 within a Regions, that means that any codec whose Rate is equal or less then 64 kbps will be alowed in that Region (G711, G729, iLBC and etc). When you set region to G729, then the codecs that are using equal or less Rate then the G729 (8kbps) will be allowed, (No G711, No iLBC any more). Now for Gatekeeper - you can use the Bandwidth command, which will make almost the same for allowing codes, For instance you can set Bandwidth Session command to 128 (G711 for GK CAC uses 2x64), so that G711 and lower Bit Rate Codecs can be used. And if you set Bandwidth Session command to 16 (G629 for GK CAC uses 2x8), then G729 and lower Bit Rate Codecs can be used. Hope that helps. On Tue, Jan 10, 2012 at 11:56 PM, Justin McIntyre justin.mcint...@blackbox.com wrote: Hello everyone. ** ** My question today is concerning controlling which codecs are used when utilizing RAS signaling via the Gatekeeper. I understand that I can control my codec inbound (at the BR2 CME) site via a inbound dial-peer that only utilizes g729r8. I also understand that using Transcoding at the same location will allow me to talk to a locally attached SIP phone (at CME site) that is configured to use g711ulaw only. However I am unclear as to how to program the CUCM controlled devices what codec to use when sourcing calls to the BR2 site via gatekeeper. If I am sourcing calls from CUCM across a gatekeeper trunk that has been configured to be in the HQ device pool which is associated with the HQ region which uses g711 intra-cluster… then shouldn’t I be sourcing packets from CUCM to BR2 as g711ulaw? Any additional thoughts or clarification would be greatly appreciated. ** ** Thanks, ** ** Justin McIntyre Engineer *Mutual Telecom Services Inc.* *a wholly-owned subsidiary of Black Box Corp.* COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 *Please note new e-mail address* *justin.mcint...@blackbox.com alex.heve...@blackbox.com *** ** ** ** ** ** ** -- This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ 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 http://www.platinumplacement.com/ image001.gifimage002.gifimage003.gif___ 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] Gatekeepers
Thanks, I just read the article and starting the INE deep dive, really helpful stuff, I recommend for anyone facing the same problem. Regards, 2012/1/10 datucha123 datucha123 datucha...@gmail.com INE Deep Dives, and Advanced Technology Class videos. They are great, Also please refer to INE Blog for Gatekeepers: http://blog.ine.com/2009/01/15/gatekeeper-call-routing-at-a-glance/ On Tue, Jan 10, 2012 at 7:12 PM, Jeferson Guardia jefers...@gmail.comwrote: Hi, I need to brush up my gwgk skills, I am really bad at it. Anyone would recommend books , videos , a particular lab? Thank you, -- Jeferson Guardia CCIE #28157 ___ 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 http://www.platinumplacement.com/ -- Jeferson Guardia CCIE #28157 ___ 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] Gatekeeper Codec control
Hi Justin, Create a new Region with 729 codec only, create new Device Pool with this region and assign your trunk to this new Device Pool. Cheers, Boris On Wed, Jan 11, 2012 at 6:56 AM, Justin McIntyre justin.mcint...@blackbox.com wrote: Hello everyone. ** ** My question today is concerning controlling which codecs are used when utilizing RAS signaling via the Gatekeeper. I understand that I can control my codec inbound (at the BR2 CME) site via a inbound dial-peer that only utilizes g729r8. I also understand that using Transcoding at the same location will allow me to talk to a locally attached SIP phone (at CME site) that is configured to use g711ulaw only. However I am unclear as to how to program the CUCM controlled devices what codec to use when sourcing calls to the BR2 site via gatekeeper. If I am sourcing calls from CUCM across a gatekeeper trunk that has been configured to be in the HQ device pool which is associated with the HQ region which uses g711 intra-cluster… then shouldn’t I be sourcing packets from CUCM to BR2 as g711ulaw? Any additional thoughts or clarification would be greatly appreciated. ** ** Thanks, ** ** Justin McIntyre Engineer *Mutual Telecom Services Inc.* *a wholly-owned subsidiary of Black Box Corp.* COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 *Please note new e-mail address* *justin.mcint...@blackbox.com alex.heve...@blackbox.com *** ** ** ** ** ** ** -- This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ 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 image003.gifimage002.gifimage001.gif___ 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] Gatekeepers
Try to get an old cisco press book called Gateways and Gatekeepers. Good source of information. On Tue, Jan 10, 2012 at 5:40 PM, Jeferson Guardia jefers...@gmail.comwrote: Thanks, I just read the article and starting the INE deep dive, really helpful stuff, I recommend for anyone facing the same problem. Regards, 2012/1/10 datucha123 datucha123 datucha...@gmail.com INE Deep Dives, and Advanced Technology Class videos. They are great, Also please refer to INE Blog for Gatekeepers: http://blog.ine.com/2009/01/15/gatekeeper-call-routing-at-a-glance/ On Tue, Jan 10, 2012 at 7:12 PM, Jeferson Guardia jefers...@gmail.comwrote: Hi, I need to brush up my gwgk skills, I am really bad at it. Anyone would recommend books , videos , a particular lab? Thank you, -- Jeferson Guardia CCIE #28157 ___ 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 http://www.platinumplacement.com/ -- Jeferson Guardia CCIE #28157 ___ 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
