hi Guys,

Thanks for your replies.

Does this also require the setup of a  call handler for *5002 and 
distribution lists in Unity Connection.

OR 

will this just work by assoicating the User B to VM Box in Unity connection?


Thanks once again!

-Vir






From: William Bell <b...@ucguerrilla.com>
Sent: Tue, 21 Aug 2012 03:00:46 
To: Dan Quinlan (daquinla) <daqui...@cisco.com>
Cc: Online Study <ccie_voice@onlinestudylist.com>, virajith 
<vir...@rediffmail.com>
Subject: Re: [OSL | CCIE_Voice] UNITY CONNECTION ISSUE! (virajith )
  Woops. That would be 0 points! Thanks for the correction.
-Bill

--William Bellblog: http://ucguerrilla.comtwitter: @ucguerrilla



On Aug 20, 2012, at 5:12 PM, Dan Quinlan (daquinla) wrote:




One correction: step 4 should say *5002. 



DQ
d...@cisco.com



Sent from my iPhone


On Aug 20, 2012, at 4:46 PM, "William Bell" <b...@ucguerrilla.com> wrote:






To expand on Vitti's excellent comments. For the latter solution, one approach 
is to create a CTI Route Point or CTI Port that is essentially a "dummy" 
device. Set the extension of that device to something generic and easy to use. 
A common example is "*XXXX".
 Set the line to CFA to voicemail. The call flow works like this:



Assume:
User A: some outside or inside caller
User B: extension 5001
User C: extension 5002



1. User A calls User B at 5001
2. User B answers
3. User A wants to leave a voicemail for user C
4. User B transfers the active call to *5001
5. The call setup for the new call leg is processed by UCM digit analysis, 
forwarding manager (since we are CFA on the CTI RP)
- Since CTI RP has extension with wildcard masks, it will match *XXXX (or 
asterisk followed by any 4 digits)
- When the call is forward to VM, the RDNIS info is set to the 4-digits









For #5 to work, you will also want a new VM profile just for the transfer. You 
want to avoid sending the asterisk (or whatever prefix/"feature code" you 
chose) in the call setup to VM.  To clarify, on the VM profile add a mask.



HTH.



-Bill




--
William Bell
blog: http://ucguerrilla.com
twitter: @ucguerrilla









On Aug 20, 2012, at 1:54 PM, rohith.ra wrote:

For transfering the call to PSTN number while in ringing state you need to 
configure idivert feature for the phone. this can be done in the softkey 
template and assign idivert button in ringing state. And assign that softkey 
template
 to the phone.



For second requirement create the specific number pattern in the directory 
number and check forward all to voicemail on that line.



-

Vitti



On Mon, Aug 20, 2012 at 11:19 PM, 
<ccie_voice-requ...@onlinestudylist.com> wrote:


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Today's Topics:



   1. Re: UNITY CONNECTION ISSUE! (virajith )





----------------------------------------------------------------------



Message: 1

Date: 20 Aug 2012 17:49:29 -0000

From: "virajith " <vir...@rediffmail.com>

To: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com>

Subject: Re: [OSL | CCIE_Voice] UNITY CONNECTION ISSUE!

Message-ID:

        
<1345442559.S.13142.19328.H.TmNjaWVfdm9pY2UtcmVxdWVzdEBvbmxpbmVzdHVkeQBDQ0lFX1ZvaWNlIERpZ2VzdCwgVm9sIDc4LCBJc3M_.RU.rfs223,

        rfs223, 724,

        
557.f5-224-130.old.1345484969.29...@webmail.rediffmail.com>

Content-Type: text/plain; charset="utf-8"





hi Guys,



A phone user should be able to divert an incoming ringing call to VM without 
answering . Also he should be able to transfer active calls on his phone to 
voicemail by dialing a specific number (mapped to VM). How should I achieve 
this requirement using Unity

connection?



Thanks,

Vir











From: ccie_voice-requ...@onlinestudylist.com

Sent: Mon, 20 Aug 2012 11:32:39

To: ccie_voice@onlinestudylist.com

Subject: CCIE_Voice Digest, Vol 78, Issue 67

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. CUCM - Routing per source extension (Isamar Maia)



   2. Re: CUCM - Routing per source extension (Dan Quinlan 
(daquinla))



   3. CUPC and Localization (Jeff S)



   4. Reply: VTGOPC.xml (Micky Grover)











----------------------------------------------------------------------







Message: 1



Date: Sun, 19 Aug 2012 13:51:32 -0300



From: Isamar Maia <isa...@gmail.com>



To: ccie_voice@onlinestudylist.com



Subject: [OSL | CCIE_Voice] CUCM - Routing per source extension



Message-ID:



   <CAPzHo3ijTfgL2RqCgcijW4a5dbVbsJjwd0YKrde==p-otc9...@mail.gmail.com>



Content-Type: text/plain; charset=ISO-2022-JP







Hi Folks,







We have here the following scenario on CUCM.







All users are dialing out through one group which has a GSM and TDM



gateway included, and both



gateways are connected using SIP.







If GSM connection fails, calls go through E1 gateway. It's working OK now.







Problem: Some users don't want to go through GSM gateway since it



doesn't have CallerID set.







So, we need to set these users to go directly to the E1 gateway.







Which are the options to the set that on CUCM conditionally based on



the call's source ?











--



Isamar Maia



Cel. VIVO SSA:  (55) 71-9146-8575



Cel. TIM SSA: (55) 71-9185-5264



Fixo:  (55) 71-4062-8688



??: +81-(0)3-4550-1212



Skype ID: isamar.maia











------------------------------







Message: 2



Date: Sun, 19 Aug 2012 17:38:36 +0000



From: "Dan Quinlan (daquinla)" <daqui...@cisco.com>



To: Isamar Maia <isa...@gmail.com>



Cc: "ccie_voice@onlinestudylist.com" 
<ccie_voice@onlinestudylist.com>



Subject: Re: [OSL | CCIE_Voice] CUCM - Routing per source extension



Message-ID: <4e55d159-c59a-4917-b24b-3ea145df8...@cisco.com>



Content-Type: text/plain; charset="iso-2022-jp"







Users who want the E1 first have a different CSS / partition where the route 
patterns point to a route list that has the route groups ordered E1 first, GSM 
second.







DQ



d...@cisco.com







Sent from my iPhone







On Aug 19, 2012, at 12:52 PM, "Isamar Maia" <isa...@gmail.com> 
wrote:







> Hi Folks,



>



> We have here the following scenario on CUCM.



>



> All users are dialing out through one group which has a GSM and TDM



> gateway included, and both



> gateways are connected using SIP.



>



> If GSM connection fails, calls go through E1 gateway. It's working OK 
now.



>



> Problem: Some users don't want to go through GSM gateway since it



> doesn't have CallerID set.



>



> So, we need to set these users to go directly to the E1 gateway.



>



> Which are the options to the set that on CUCM conditionally based on



> the call's source ?



>



>



> --



> Isamar Maia



> Cel. VIVO SSA:  (55) 71-9146-8575



> Cel. TIM SSA: (55) 71-9185-5264



> Fixo:  (55) 71-4062-8688



> ??: +81-(0)3-4550-1212



> Skype ID: isamar.maia



> _______________________________________________



> 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: 3



Date: Sun, 19 Aug 2012 11:07:35 -0700



From: "Jeff S" <jes20...@gmail.com>



To: "'Online Study'" <ccie_voice@onlinestudylist.com>



Subject: [OSL | CCIE_Voice] CUPC and Localization



Message-ID: <000001cd7e35$83fc9f30$8bf5dd90$@gmail.com>



Content-Type: text/plain; charset="us-ascii"







All,







I have CUCM setup to globalize and localize my incoming calls. This is



working great on the hard phones, but not on CUPC.  















When I dial 7775002 on my PSTN phone, this call is displayed on the HQ Phone



2 hard phone as 7773434 and as 
+14087774343 in the Missed Calls directory.



This is working as I'd expect.















However, when using CUPC in deskphone mode, the number is displayed as



+14087774343 when the call is ringing in(and in CUPC Missed Calls



directory).  When I move to softphone mode, call is then presented as



7774343 while ringing and in the Missed Calls directory.  















Removing the Calling Party Transformation CSS from the phone doesn't change



the deskphone mode results, but it does "unlocalize" the softphone mode



results to where both the display and Missed Calls directory both display



+14087774343.























My question is: How can we configure CUPC for globalization/localization



just like an IP Phone?















--j















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------------------------------







Message: 4



Date: Mon, 20 Aug 2012 11:31:38 +0530



From: Micky Grover <mickygrove...@gmail.com>



To: ccie_voice@onlinestudylist.com



Subject: [OSL | CCIE_Voice] Reply: VTGOPC.xml



Message-ID:



   <cagxxwivmsetypaq6ugql8bhqkn9hcczthcyn1ug1qvq-mp9...@mail.gmail.com>



Content-Type: text/plain; charset="iso-8859-1"







Hey Amaia,







You can check this link for more info. about this product::







http://www.ipblue.com/download/products/pc/2.10.1.120/readme.txt







Thanks:



Mohit Grover















On Sun, Aug 19, 2012 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. Rack - Unable to register PSTN phone using IP 
    blue as 7960



>       (Amaia Lesta)



>    2. Phoneview - CME (Randall Crumm)



>    3. Re: Phoneview - CME (Kevin Spicer)



>



>



> ----------------------------------------------------------------------



>



> Message: 1



> Date: Sat, 18 Aug 2012 20:48:00 +0200



> From: Amaia Lesta <amaia.le...@gmail.com>



> To: ccie_voice@onlinestudylist.com



> Subject: [OSL | CCIE_Voice] Rack - Unable to register PSTN phone using



>         IP     
 blue as 7960



> Message-ID:



>         <



> 
cak8ojdb5dc-pfyaxx+tdvm8k2qtw0q3cpj2jksrxbv_14np...@mail.gmail.com>



> Content-Type: text/plain; charset="iso-8859-1"



>



> Hello all,



>



> After many sessions trying I am unable to register IP Blue 7960 as PSTN



> phone.



> It gets stuck registering in the middle of the TFTP process searching 
for



> VTGOPC.xml and VTGO.cfg.xml



>



> I configure as TFTP server the PSTN WAN ROUTER (10.10.100.2).



> MAC address the one of the MS loopback adapter 02004C4F4F50



> Phone type 7960



> And check the option "Use Cisco VPN client". => The softphone 
can't see any



> network adapter in my PC, so I can't select it.



>



> This is the output of the TFTP debug in the router:



>



> Aug 18 22:32:21.675: TFTP: Looking for SEP02004C4F4F50.cnf.xml



> Aug 18 22:32:21.675: TFTP: Opened system:/its/XMLDefault7960.cnf.xml, 
fd 0,



> size 933 for process 327



> Aug 18 22:32:22.007: TFTP: Finished system:/its/XMLDefault7960.cnf.xml,



> time 00:00:00 for process 327



> Aug 18 22:32:22.175: New Skinny socket accepted [2] (0 active)



> Aug 18 22:32:22.175: sin_family 2, sin_port 51845, in_addr 10.10.0.76



> Aug 18 22:32:22.175: skinny_add_socket 2 10.10.0.76 51845



> Aug 18 22:32:22.195: TFTP: Looking for VTGOPC.xml



> Aug 18 22:32:22.359: TFTP: Looking for VTGO.cfg.xml



>



>



> And the ephone exists in the system



> ephone  1



>  device-security-mode none



>  mac-address 0200.4C4F.4F50



>  type 7960



>  button  1:1 2:2 3:3 4:4



>  button  5:5 6:6



> ephone  2



>



> Can anyone help me on this one?



>



> Thanks for your support :)



>



> Amaia



> -------------- next part --------------



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>



> ------------------------------



>



> Message: 2



> Date: Sun, 19 Aug 2012 06:34:25 -0700 (PDT)



> From: Randall Crumm <rrcr...@yahoo.com>



> To: Online Study <ccie_voice@onlinestudylist.com>



> Subject: [OSL | CCIE_Voice] Phoneview - CME



> Message-ID:



>         
<1345383265.2316.yahoomail...@web124904.mail.ne1.yahoo.com>



> Content-Type: text/plain; charset="iso-8859-1"



>



> Hello,



> I set up phoneview for cme as per the document on proctorlabs. I was 
able



> to test and import the 2 phones, but I cannot control either? I see 
x4001



> and 4002 on the banner but I have 
+442077964001 and 2 respectively.



>



>



>



> Any thoughts?



> ?



> Cheers,



> Randall



> -------------- next part --------------



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>



> ------------------------------



>



> Message: 3



> Date: Sun, 19 Aug 2012 15:55:28 +0100



> From: Kevin Spicer <ke...@kevinspicer.co.uk>



> To: Randall Crumm <rrcr...@yahoo.com>



> Cc: Online Study <ccie_voice@onlinestudylist.com>



> Subject: Re: [OSL | CCIE_Voice] Phoneview - CME



> Message-ID:



>         <



> 
caf2ggot5rwb+kvckujby45iovxvtdef9k3v9uxnh7dwxn7i...@mail.gmail.com>



> Content-Type: text/plain; charset="iso-8859-1"



>



> Did you specify the phone type in the ephone - its required.



>



> On 19 Aug 2012 14:34, "Randall Crumm" 
<rrcr...@yahoo.com> wrote:



>



> > Hello,



> > I set up phoneview for cme as per the document on 
proctorlabs. I was able



> > to test and import the 2 phones, but I cannot control either? 
I see x4001



> > and 4002 on the banner but I have 
+442077964001 and 2 respectively.



> >



> >



> >



> > Any thoughts?



> >



> > Cheers,



> > 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



> >



> -------------- next part --------------



> An HTML attachment was scrubbed...



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> 
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>



> ------------------------------



>



> _______________________________________________



> CCIE_Voice mailing list



> CCIE_Voice@onlinestudylist.com



> 
http://onlinestudylist.com/mailman/listinfo/ccie_voice



>



>



> End of CCIE_Voice Digest, Vol 78, Issue 66



> ******************************************



>















--



*Thanks:*



*Mohit Grover*



MCP, MCSE, MCITP



CCIE Voice # 35961



BCA, MCA (Network Communications) (Wireless/Mobility)



MBA (Business Strategies)







Everybody is a GENIUS, But if you judge a fish by its ability to climb a



tree, it will live its whole life believing that it is stupid.



-Albert Einstein



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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







_______________________________________________

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 
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Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

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