Re: [OSL | CCIE_Voice] BACD with the latest IOS versions
Hi, You can safely ignore the warning. The script works anyway. Sent from my mobile device, sorry for typos. --- Regards Boris On 04/03/2012, at 9:17, Emanuel Damasceno wrote: > Since you guys brought this up, I was wondering this: > Why can't I make B-ACD work here? I see I have it here inside my applications > BR2-RTR(config-app)#service ? > dsapp > ipsla-responder > clid_authen > clid_col_npw_npw > AFW_THIRD_PARTY_CC > CALLIndSs_SErviCe > Default > RetrProxy > CTAPP > clid_authen_col_npw > fax_hop_on > ipsla-testcall > app-b-acd-aa > clid_authen_npw > session > app-b-acd > clid_authen_collect > clid_col_npw_3 > WORD Name of the service/package > > > BR2-RTR(config-app)#service > > So, now I go... > > BR2-RTR(config-app-param)#param ? > Parameters registered under app-b-acd namespace: > name type default value description > uid-len I 10 the number of digits in UID > warning-time I 30 the time (in secs) within which > a user is warned before the calling time expires (call terminates) > pin-len I 4 the number of digits in PIN > retry-count I 3 the number of attempts to > reenter PIN > redirect-number S the telephone number where a > call is redirected to > WORD Parameter name > > > BR2-RTR(config-app-param)#param > > This is what happens... > BR2-RTR(config-app)#service app-b-acd-aa > BR2-RTR(config-app-param)#paramspace english index 0 > BR2-RTR(config-app-param)#param max-time-call-retry 700 > Warning: parameter max-time-call-retry has not been registered under > app-b-acd-aa namespace > > What should I do to make B-ACD work? > Emanuel Damasceno > CCNP Voice > > > > > > On Sat, Mar 3, 2012 at 6:19 PM, Rajasekar Shanmugam > wrote: > Thanks a lot Vik. This helps. > > Raj > > > On Sat, Mar 3, 2012 at 2:18 PM, Vik Malhi wrote: > You can see which applications are running as shown below (and yes is the > answer to your question). > > SiteC-RTR#sh call application voice summ > > SERVICES (standalone applications): > name typedescription > > dsapp C Scriptbuiltin:DSESS_Service.C > ipsla-responder Tcl Script builtin:app_test_rcvr_script.tcl > clid_authen Tcl Script > builtin:app_clid_authen_script.tcl > clid_col_npw_npw Tcl Script > builtin:app_clid_col_npw_npw_script.tcl > AFW_THIRD_PARTY_CCC Scriptbuiltin::Third_Party_CC_Service.C > > CALLIndSs_SErviCe C Scriptbuiltin:CallIndSs_Service.C > Default C Scriptbuiltin:Session_Service.C > CTAPP C Scriptbuiltin:CallTreatment_Service.C > clid_authen_col_npw Tcl Script > builtin:app_clid_authen_col_npw_script.tcl > fax_hop_onTcl Script builtin:app_fax_hop_on_script.tcl > > ipsla-testcallTcl Script builtin:app_test_place_script.tcl > > app-b-acd-aa Tcl Script builtin:app_b_acd_aa_script.tcl > clid_authen_npw Tcl Script > builtin:app_clid_authen_npw_script.tcl > session Tcl Script builtin:app_session_script.tcl > app-b-acd Tcl Script builtin:app_b_acd_script.tcl > clid_authen_collect Tcl Script > builtin:app_clid_authen_collect_script.tcl > clid_col_npw_3Tcl Script > builtin:app_clid_col_npw_3_script.tcl > > > Vik Malhi – CCIE #13890 > Managing Partner - IPexpert, Inc. > > Telephone: +1.810.326.1444 ext 420 > Fax: +1.810.454.0130 > Mailto: vma...@ipexpert.com > > > > > On Mar 2, 2012, at 1:31 PM, Rajasekar Shanmugam wrote: > >> Experts - >> >> I would like to know , if the BACD application is supported without the TCL >> scripts in the flash. Meaning , is there an embedded application / script >> available with the later IOS releases ? >> -- >> Raj >> ___ >> 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 > > > > > -- > Raj > > ___ > For more information regarding industry leading
Re: [OSL | CCIE_Voice] MRGL order
Yes, RSVP MTPs should be the last choice in the list so that they are not used as regular MTPs. As for the CUCM MTPs, they support 711 codecs only. If lab to not restrict their usage and it suits other requirements, I'd go with this option. On Tue, Feb 28, 2012 at 4:18 PM, AJ BG wrote: > Hello All, > > If you have multiple resources with overlapping functionality then what is > the correct MRG order? > Let say there are MTP hardware, Server MTP, Transcoding, and RSVP > I would configure it this way > MRGL-HQ > 1. MRG-MTP-Hardwar > 2. MRG-MTP-Servers > 3. MRG-Xcoding-HQ > 4. MRG-RSVP-HQ > Is this the correct order? I don’t want my transcoding or RSVP used as > MTP. > If you need to invoke a MTP resources, but there is no preference in the > workbook, is it ok to use the servers’ MTP? Or should we create a MTP > resource in the router? > Thanks, > AJ > > ___ > 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] overlapping dial-plan
Just use 9[1-9]... instead. On Tue, Feb 28, 2012 at 4:35 PM, AJ BG wrote: > Hello All, > > In the IOS gateway how would you solve an overlapping dial-plan issue > between 8 digit European dialing with access code of 9 “9……..”and > international pattern “900T”. Assuming that the LD plan that is given to > you is 9 followed by 8digits. > > The router will send the call as soon as it has enough digits. Is there > really anyway to achieve this, beside configure your LD as 9[1-9]…….. > > Thanks, > > AJ > > > ___ > 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
Afaik you never need multicast routing configured for the Moh from flash because this traffic has TTL=1, so it wont traverse Layer 3 boundary. This is why you need the route attribute in the moh statement. Sent from my mobile device, sorry for typos. --- Regards Boris On 15/02/2012, at 8:28, Emanuel Damasceno wrote: > Hello Experts, > > I am watching IPExpert's video on demand on Multicast MOH. > > Did I understand this right? > > "multicast moh 239.1.1.1 port 16384" < if I have it like this, I have to put > "ip pim dense-mode" on my interfaces > > "multicast moh 239.1.1.1 port 16384 route 10.10.201.1 10.10.110.2" < if I > have it like this, I DON'T have to put "ip pim dense-mode" on my interfaces > > Could somebody please confirm? > Thanks > Emanuel Damasceno > CCNP 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 ___ 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] CUPS 7.0.1
Hi mate, In my home lab I use the latest available on CCO CUCM/CUC/CUPS/UCCX 7.0 with demo licenses. It is all on VMs, so when license expires for CUPS or UCCX I simply roll back to postinstall state. The only feature missing in the demo licenses is VPIM networking for CUC. As for your issue with information update, make sure that you have CUPS ip address in Application Server settings in CUCM. And that the versions you use are compatible according to documentation. Sent from my mobile device, sorry for typos. --- Regards Boris On 31/01/2012, at 7:24, Justin McIntyre wrote: > Good Day everyone, > > My Question today is in regards to the Presence Server > interoperability with UCM 7.0.1. I have CUPS 7.0.0 and it turns out that > there is a good difference in function between CUPS 7.0.1 and CUPS 7.0.0. I > have the 8.0 NFR kit and did some testing with the integration of CUCM 7.0.1 > and CUPS 8.0.2 however I am seeing some difficulties in getting the system to > update user information across from UCM towards CUPS. I.E. I am having to > reset CUPS to get it to resync information that I have updated on the CUCM > side of the integration. I’m sure a lot of that has to do with my experience > and knowledge level of the software release upon which I am working. What I > am looking for is if anyone has used CUPS 8.0 in place of 7.0.1 for their > CCIE Voice studies. I have not been able to find the NFR software for 7.0.1. > It seems that it has been discontinued for sale. I was able to come up with > all the other required software other than CUPS which as I stated I have > 7.0.0.9000xxx. Thank you in advanced for the help. > > Thanks, > > Justin > > > 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 ___ 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] IPX vol1 Lab9a
hi mate, you can get IPMA and EM URLs from the CUCM Help page, just search for DEVICENAME UCCX Agent Login and IPPM URLs you need to memorise or remember shortcuts on Cisco site. Regards, Boris On Wed, Jan 25, 2012 at 1:34 PM, Emanuel Damasceno wrote: > No brother... > > The best way is to find and learn where it is on Cisco DocCD... :) > *Emanuel Damasceno* > CCNP Voice > > > > > > On Wed, Jan 25, 2012 at 12:09 AM, Jeferson Guardia wrote: > >> IPMA complex, too many steps.. but somehow I feel after practicing >> for a while I will get used to it. IPX covers well but for some steps, even >> watching the video, it doesnt take you where you need a piece of specific >> information to accomplish a task, example: >> >> Adding IP Phone Services for IPMA and EM (Extension Mobility) , I know >> how to create but the URL string I should create, it doesnt mention where >> to find it or the full string for it. After googling it, I found the URL. >> Anyone would have a better way to find these URL`s in case in the future I >> have to deal with adding new IP Phone Services? This was new to me. >> >> Thank you all!! >> >> -- >> 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 > ___ 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] Corlist not blocking intl call
Hi mate, Most likely your mobile dialpeer is letting the call out. Sent from my mobile device, sorry for typos. --- Regards Boris On 23/01/2012, at 14:38, chase mergenthal wrote: > I'm trying to block an outbound call to international; with the following > config, however my call completes... > > Any ideas? > > SiteC-RTR#sho run | sec dial-peer > dial-peer cor custom > name local > name mobile > name intl > dial-peer cor list PT-LOCAL > member local > dial-peer cor list PT-MOBILE > member mobile > dial-peer cor list PT-INTL > member intl > dial-peer cor list CSS-LOCALMOBILE > member local > member mobile > > dial-peer voice 1 pots > translation-profile incoming DID > incoming called-number . > direct-inward-dial > > dial-peer voice 1000 voip > destination-pattern [15]... > session protocol sipv2 > session target ipv4:10.10.210.11 > > dial-peer voice 1001 voip > destination-pattern [15]... > session protocol sipv2 > session target ipv4:10.10.210.10 > > dial-peer voice 10 pots > corlist outgoing PT-INTL > translation-profile outgoing INTL > destination-pattern 900T > port 0/0/0:15 > prefix 00 > > dial-peer voice 7 pots > corlist outgoing PT-LOCAL > translation-profile outgoing local > destination-pattern 97...T > port 0/0/0:15 > prefix 0207 > > dial-peer voice 9 pots > corlist outgoing PT-MOBILE > translation-profile outgoing LDMOB > destination-pattern 90..T > port 0/0/0:15 > prefix 0 > SiteC-RTR# > > > SiteC-RTR#sho run | sec ephone > max-ephones 10 > ephone-dn 1 > number 3001 no-reg primary > name Ben Bernanke > ephone-dn 2 > number 3006 no-reg primary > name George Bush > corlist incoming CSS-LOCALMOBILE > ephone 1 > device-security-mode none > mac-address 001B.D4C6.C1D9 > type 7960 > button 1:1 > ephone 2 > device-security-mode none > mac-address ..0003 > button 1:2 > SiteC-RTR# > > > -Chase > > -- > If winners never quit and quitters never win, then who coined the phrase, > "Quit while you’re still ahead."? > > ___ > 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] Gatekeeper VIAZONE - Could not find an IPIPGW
Hi Steve,
Do you have This in your config?
Voice service voip
Allow h323 to h323
If not, add it and do no gateway/gateway
Your cube should appear as H323 type in show gatek end.
Sent from my mobile device, sorry for typos.
---
Regards
Boris
On 18/01/2012, at 7:22, Steven wrote:
> Hi there,
> i got some problems with my viazone ("CUBE") at HQ-RTR.
> I already checked the Tech prefix match and it seems to succeed.
> But i'm clueless how to debug/resolve the "Could not find an IPIPGW"-problem.
> I also checked the dial-peers on HQ-RTR.
>
> Any help appreciated.
>
> Regards,
> Steven
>
>
> ! *** Begin tech details:
>
> HQ-RTR#debug gatekeeper main 10! tried to call from HQ (5002) to BR2
> (3006)
>
> Jan 17 19:56:37.786: ////GK/gk_process: QUEUE_EVENT
> (minor 0) wakeup
> Jan 17 19:56:37.786: ////GK/gk_rassrv_arq:
> arqp=0x48F0C08C,crv=0x7, answerCall=0
> Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: ARQ
> Didn't use GK_AAA_PROC
> Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No Name
> servers
> Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo:
> (3006) Tech-prefix match failed.
> Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo:
> (3006) Matched zone prefix 3 and remainder 006
> Jan 17 19:56:37.786:
> ////GK/gk_rassrv_get_ingress_network: returning
> default ingress network = 1
> Jan 17 19:56:37.786:
> //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the
> source side, src_zonep=0x4793079C
> Jan 17 19:56:37.786:
> //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is
> UCM, and z_invianamelen=0
> Jan 17 19:56:37.786:
> //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the
> destination side, dst_zonep=0x47930A08
> Jan 17 19:56:37.786:
> //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is
> UCME, and z_outvianamelen=4
> Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone
> and z_outvianamep=CUBE
> Jan 17 19:56:37.786:
> //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ for a
> zone (UCME) that has an outviazone (CUBE) specified. Pick an IP-IP gateway
> in that viazone.
> Jan 17 19:56:37.786:
> ////GK/gk_gw_select_ipipgw_random: zonep: 0x47930C74,
> tpp: 0x0, current_endpt: 0
> Jan 17 19:56:37.786:
> ////GK/gk_gw_select_ipipgw_random: Selecting any
> IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0
> Jan 17 19:56:37.786:
> ////GK/gk_gw_select_ipipgw_random: qelemp=0x46F0FE88,
> loop_count=0
> Jan 17 19:56:37.786:
> ////GK/gk_gw_select_ipipgw_random: Examining tgwp
> 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1
> Jan 17 19:56:37.786:
> ////GK/gk_gw_select_ipipgw_random: Searched through
> the entire gateway list. loop_count=0
> Jan 17 19:56:37.786:
> ////GK/gk_gw_select_ipipgw_random: Could not find an
> IPIPGW.
> Jan 17 19:56:37.786:
> //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone gateway
> selection failed for zone "CUBE"
> Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq:
> rassrv_get_addrinfo() failed (return code = 0x805)
>
>
> HQ-RTR#show gatekeeper gw-type-prefix
> GATEWAY TYPE PREFIX TABLE
> =
> Prefix: 1#*(Default gateway-technology)
> Zone CUBE master gateway list:
>10.10.110.1:1720 HQ-RTR
> Zone UCM master gateway list:
>10.10.210.10:44248 gk-trunk_1
>10.10.210.11:36641 gk-trunk_2
>
> Prefix: 3#*
> Zone UCME master gateway list:
>10.10.110.3:1720 BR2-RTR
>
>
> HQ-RTR#show gatekeeper zone prefix
> ZONE PREFIX TABLE
> =
> GK-NAME E164-PREFIX
> --- ---
> UCME 3...
> UCM 5...
>
>
> HQ-RTR#show running-config interface loopback 0
> interface Loopback0
> ip address 10.10.110.1 255.255.255.255
> h323-gateway voip interface
> h323-gateway voip id CUBE ipaddr 10.10.110.1 1719
> h323-gateway voip h323-id HQ-RTR
> h323-gateway voip tech-prefix 1#
>
>
> HQ-RTR#show running-config | section gatekeeper
> gatekeeper
> zone local UCM ipexpert.com
> zone local UCME ipexpert.com outvia CUBE
> zone local CUBE ipexpert.com
> zone prefix UCME 3...
> zone prefix UCM 5...
> gw-type-prefix 1#* default-techno
Re: [OSL | CCIE_Voice] RSVP and Multicast MoH
AFAIK MMoH will work together with RSVP because it is not taken into consideration for CAC. The multicast streams flow around MTP/RSVP reservations, but you need to cater for this traffic in priority queue towards the stream destination. Sent from my mobile device, sorry for typos. --- Regards Boris On 12/01/2012, at 5:31, datucha123 datucha123 wrote: > Hello, > > When using RSVP CAC between HQ and Branch 1 Sites, the Multicast MoH should > not work for Branch 1 IP Phones from UCM Server, as the Muticast cannot > traverse the MTP, right? > > But when sourcing Mutlicast MoH from the Branch Router, it will work. > > > ___ > 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] 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 *** > > ** ** > > ** ** > > ** ** > > -- > 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 > <><><>___ 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 wrote: > 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] vol 2 lab 3 : RSVP just won't work
Hi mate, have you resolved the issue? I am having the same problem, can't figure out why. Thanks, Boris ___ 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
