[SIPForum-discussion] Unified voicemail

Fri Dec 14 13:24:58 UTC 2007

Correct,

  Thanks,
  Biswa,
  SSE.

Rohin Koul <rohin.koul at gmail.com> wrote:
    Hi Biswa,

  Yes you are right Orig S-CSCF "can" send the INV directly to term S-CSCF bypassing the AS but this will be the case if orig subscriber doesn't have any services subscribed.
  For example if orig subscriber has "outgoing call screening" service then orig S-CSCF will trigger the AS.
  AS will in turn determine if it has to allow the call and act accordingly.

  regs

  Rohin

  On 13/12/2007, biswanath das <sam_bisu at yahoo.co.in> wrote:     Hi Rohin,
  I am having some doubts for steps 4 and 5.
  As it is cleared that, terminating AS is responsible for forwarding the call to VM, so Orig S-CSCF can transfer the call directly to Term S-CSCF, instead of sending it to Orig AS.

  Then, we can proceed from step 6.
  Please let me know, if I am going wrong.

  And Joel,
  Unified Messaging may support Voice mail, Fax, e-mail, sms etc.
  So, you can configure any UM server supporting any of these services.
  Configure any subsriber under this UM server domain,
  provided this UA is not registered.

  Then try to establish a call/ or send a sms, send a e-mail 
  to this user using x-lite or SJ phone etc.
  As this UA is not rregistered, It will automatically divert the call to this UM server.

  I think, this can help you u in UM setup.

  Thanks and Regards,
  Biswa,
  SSE.
    \Joel Silva <joel.silva at novabase.pt> wrote:

      Thanks for the reply Rohin!
  Please consider this scenario:

  A GSM or PSTN user tries to call another user (Sip or not). The user doesn't pick up and so the call is transfered for the voicemail. 
  When the message is recorded, the user hang-up and the message is sent to the user e-mail.
  What are the components that you see that I need to implement such a scenario? Please consider a non-IMS network.
  Thanks,

    Joel Silva
Analyst
  .............................................................................................................
  Novabase
Av. Eng. Duarte Pacheco, 15F . 1099-078 Lisboa - Portugal
Tel. (+351) 213 836 300 . Fax (+351) 213 836  301 .
mailto:joel.silva at novabase.pt
www.novabase.pt

---------------------------------
  From: discussion-bounces at sipforum.org [mailto: discussion-bounces at sipforum.org] On Behalf Of Rohin Koul
Sent: segunda-feira, 10 de Dezembro de 2007 13:15
To: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Unified voicemail

  I am not familiar with oracle products so I won't comment on that.
  Typically in an IMS network voice mail service may be implemented as follows

  1. a non IMS subscriber is connected through a gateway and originates a call.
  2. call arrives at gateway and gatway sends an INV to Proxy-CSCF
  3. Proxy sends the call to S-CSCF
  4. S-CSCF based on its filter criteria will send an INV to SIP-AS (if orig subscriber has any services subscribed)
  5. SIP-AS shall send the INV to terminating S-CSCF 
  6. Terminating S-CSCF will send INV to its SIP-AS and SIP-AS will send the INV back to S-CSCF (SIP-AS is B2BUA)
  7. Terminating S-CSCF will send INV to terminating gateway/subscriber
  8. If the terminating subscriber is busy a 486 is returned.
  9. 486 travels back to SIP-AS via S-CSCF
  10. On reciept of 486 SIP-AS will not send 486 to originating user but would divert the call to a Voice Mail server which would play the appropriate announcement and record a message etc.

  in short SIP-AS shall act as a B2BUA and do necessary signalling to divert the call to VM server.
  SIP-AS may also implement other variants of voicemail service like CFVMNR, CFVM, CFVMB etc

  On 10/12/2007, Joel Silva <joel.silva at novabase.pt > wrote:       For now, I'm not trying to do this in an IMS network.. What are the main advantages of doing this in an IMS network?
  I use the Oracle Communication and Mobility Server as the SIP AS.

  Thanks,

    Joel Silva
Analyst
  .............................................................................................................
  Novabase
Av. Eng. Duarte Pacheco, 15F . 1099-078 Lisboa - Portugal
Tel. (+351) 213 836 300 . Fax (+351) 213 836  301 .
mailto:joel.silva at novabase.pt
www.novabase.pt

---------------------------------
  From: discussion-bounces at sipforum.org [mailto: discussion-bounces at sipforum.org] On Behalf Of Rohin Koul
Sent: segunda-feira, 10 de Dezembro de 2007 12:06
To: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Unified voicemail

  Two more things I can think of are
  1. Voice Mail server
  2. SIP aplication server

  btw are you trying to do this in an IMS network ??

  On 10/12/2007, Joel Silva <joel.silva at novabase.pt > wrote:       Hello all!
  Can someone give me some hints about the necessary requirements to build a unified voicemail?
  The ones that I identify until now:
  1. PSTN Gateway
  2. GSM Gateway

  Thanks,

    Joel Silva
Analyst
  .............................................................................................................
  Novabase
Av. Eng. Duarte Pacheco, 15F . 1099-078 Lisboa - Portugal
Tel. (+351) 213 836 300 . Fax (+351) 213 836  301 .
mailto:joel.silva at novabase.pt
www.novabase.pt

---------------------------------
  From: discussion-bounces at sipforum.org [mailto: discussion-bounces at sipforum.org] On Behalf Of Ritambhara Mittal
Sent: segunda-feira, 10 de Dezembro de 2007 10:33
To: Ritambhara Mittal; nidhi jain; discussion at sipforum.org
Subject: [SIPForum-discussion] need information on R3protocol specs.????

Ritambhara Mittal <ritambhara.mittal at yahoo.co.in> wrote:     Hey, 
  Do anyone know about R3 protocol specs.

  Thanks

nidhi jain <nidhi_jain6680 at yahoo.co.in> wrote:
    what is diversion...??
  how diversion and replace parameter differ.

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