Hi guys,
I am trying to achieve the following with UCCX : 1)users should hear a script saying "Thank u for calling" and "All our reps are busy at this time please stay on the line someone will be with u shortly " 2) The script should also be capable of playing to the caller his position in queue such as "Your Position is Y" where Y stands for position of the call in the q Questions: 1) how do I go about achieving this? 2)how does one get such a script ? Does it need to be seperately recorded and how? 3)what steps need to performed for this configuration on UCCX? Thanks, Vir From: ccie_voice-requ...@onlinestudylist.com Sent: Sun, 16 Sep 2012 12:28:11 To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 79, Issue 49 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 trunk (John John) 2. Plan and Type fields (Bruno Nonogaki) 3. Re: Plan and Type fields (Marcelo Alexandria) 4. Unity - Prompt Recording With Phone (nehal ahmed) ---------------------------------------------------------------------- Message: 1 Date: Sat, 15 Sep 2012 09:13:44 -0700 (PDT) From: John John <john_ccie2...@yahoo.com> To: Randall <rrcr...@yahoo.com>, Bill Lake <whl...@gmail.com> Cc: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com> Subject: Re: [OSL | CCIE_Voice] sip trunk Message-ID: <1347725624.64869.yahoomail...@web122606.mail.ne1.yahoo.com> Content-Type: text/plain; charset="iso-8859-1" Dears, ? ?In my case which I have only the voice gateways and the customers have their own PBX so I think the only way to do that is only using incoming uri command under the dial-peer which will match the ip for the cutomer side and then I will apply translation profile on that dial peer and it should work so tomorrow I will test it and I will let you know. ? Regards ? ________________________________ From: Randall <rrcr...@yahoo.com> To: Bill Lake <whl...@gmail.com> Cc: John John <john_ccie2...@yahoo.com>; "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com> Sent: Saturday, September 15, 2012 5:50 AM Subject: Re: [OSL | CCIE_Voice] sip trunk Did you configure the cucm server up Address on the cme? Randall On Sep 14, 2012, at 7:32 PM, Bill Lake <whl...@gmail.com> wrote: Because they are coming in on a seperate sip dial peer, but maybe that won't work. ?I would think it wouldo because you would only apply the change inbound from them ( you said they are changing ani in their switch) and only allow the ani you want > >On Friday, September 14, 2012, John John wrote: > >I get what you are talking about but this is not my case. >>? >>let me explain again: >>? >>we have one voice gateway with 2 E1 for 2 customers A,B >>and the connection between us and customer A,B is sip trunk so they have their own PBX and I have already assigned range of DID for both company and as you know I have 2 dial-peer between me and the customer PBX. >>? >>dial-peer voice 1 voip >>description ## Customer-A ## >>answer-address 332211.. >>destination-pattern 332211.. >>modem passthrough nse codec g711ulaw >>session protocol sipv2 >>session target ipv4:192.168.10.10 >> >>dial-peer voice 2 voip >>description ## Customer-B ## >>answer-address 332222.. >>destination-pattern 332222.. >>modem passthrough nse codec g711ulaw >>session protocol sipv2 >>session target ipv4:192.168.20.10 >> >> >>so now for ougoing calls from customer A the call path will be?? Customer A PBX---------Mygateway------PSTN over E1 >>so If customer A send the calling number as 33222222 which is customer B range then the call will go with customer B DID so how can I block them. >> >> >> From: Bill Lake <whl...@gmail.com> >>To: John John <john_ccie2...@yahoo.com> >>Cc: Pavan <pav.c...@gmail.com>; "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com> >>Sent: Saturday, September 15, 2012 1:11 AM >>Subject: Re: [OSL | CCIE_Voice] sip trunk >> >> >>Sure you can, you know that any call incoming from Site A must have ANI of 332211 and any call incoming from Site B must have 332222 so you just need to enforce that and you could do that with a incoming voice translation rule.? >> >>So no matter what the customer sends you can change that to what you want it to be >>? >>We do this all the time in the work books >> >>voice translation-rule 1 >>rule 1 /^3...$/ /408387\0/ >> >>this expands 3001 to 4083873001 >> >>but you could also do >> >>rule 1 /^3...$/ /4001/? >> >>This would take any 3XXX number and change it to 4001, why you ask, just think you have 800 people calling out from your company call center and this prevents 800 different numbers from going out. Everyone you call gets 4001.? not 3001-3799 or something like that. >> >>You get the idea? >> >> >> >> >>On Fri, Sep 14, 2012 at 4:35 PM, John John <john_ccie2...@yahoo.com> wrote: >> >>Dear Pavan, >>>? >>>Thanks for your feedback,But my question is even if i implement cor still I can't control that because I can't control what customer A and B and in that case if customer A sent me the call with ANI for cutomer B then customer B dial-peer will be match and the call will not be blocked. >>>? >>>but during my google search I found some greating feature which is incoming uri based on the IP so anybody working in that feature????? >>> >>>? >>> From: Pavan <pav.c...@gmail.com> >>>To: John John <john_ccie2...@yahoo.com> >>>Cc: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com> >>>Sent: Friday, September 14, 2012 6:43 AM >>>Subject: Re: [OSL | CCIE_Voice] sip trunk >>> >>> >>>I think cor can offer a good solution for this. Assign a separate cor group to each dialpeer say cust1 and cust2 in both inbound and outbound directions >>> >>>-Pavan >>> >>>On Sep 13, 2012, at 17:45, John John <john_ccie2...@yahoo.com> wrote: >>> >>> >>>Dear All, >>>> >>>>?We have two sipt?trunk for 2 comapany:- >>>> >>>>company A - DID range 332211XX >>>>Company B - DID range 332222XX >>>> >>>>and each company has own PBX,Company A? has AVAYA and company B has cisco Call manager. >>>> >>>>and they have sip trunk to my gateway where is the E1 is connected. >>>> >>>>in my gateway there is two dial peer : >>>> >>>>dial-peer voice 1 voip >>>>?description ## Customer-A ## >>>>?answer-address 332211.. >>>>?destination-patt _______________________________________________ >For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ > >Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: </archives/ccie_voice/attachments/20120915/d8ef38be/attachment-0001.html> ------------------------------ Message: 2 Date: Sat, 15 Sep 2012 15:19:11 -0300 From: Bruno Nonogaki <brun...@gmail.com> To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Plan and Type fields Message-ID: <cap_rldv6tfp6fhc04zla0_tp1k5pp8v0mdpwdzh_azn8x6y...@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hey guys, If the question asks me to mark a Called Party Number *Type* as Subscriber/Local/International, but does not say anything about Called Party Numbering *Plan*, am I supposed to mark it with ISDN also? Thank you, Bruno -------------- next part -------------- An HTML attachment was scrubbed... URL: </archives/ccie_voice/attachments/20120915/370e0c9e/attachment-0001.html> ------------------------------ Message: 3 Date: Sat, 15 Sep 2012 17:10:51 -0300 From: Marcelo Alexandria <malexand...@uol.com.br> To: Bruno Nonogaki <brun...@gmail.com> Cc: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com> Subject: Re: [OSL | CCIE_Voice] Plan and Type fields Message-ID: <19e82ac8-83a6-493d-a021-8d213fdf6...@uol.com.br> Content-Type: text/plain; charset=utf-8 Yes Enviado via iPhone Desculpe erros de digita??o Em 15/09/2012, ?s 15:19, Bruno Nonogaki <brun...@gmail.com> escreveu: > Hey guys, > > If the question asks me to mark a Called Party Number Type as Subscriber/Local/International, but does not say anything about Called Party Numbering Plan, am I supposed to mark it with ISDN also? > > Thank you, > > 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 ------------------------------ Message: 4 Date: Sun, 16 Sep 2012 11:48:27 +0500 From: nehal ahmed <nehal.ah...@msn.com> To: <ccie_voice@onlinestudylist.com> Subject: [OSL | CCIE_Voice] Unity - Prompt Recording With Phone Message-ID: <snt144-w10d08e38e5f54e716f26b8e0...@phx.gbl> Content-Type: text/plain; charset="iso-8859-1" Hi, I am using proctorslab and unable to do prompt recording when selecting "Phone" as a recording device. I don't receive any call on the specified Extension for Recording.However it works fine when selected for "Computer" , Record and Play. Can anyone guide me whats the problem is ? Regard Nehal -------------- next part -------------- An HTML attachment was scrubbed... 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_______________________________________________ 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
