[SIPForum-discussion] Unified voicemail

Thu Dec 13 08:23:16 UTC 2007

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 <joel.silva at novabase.pt>
> *www.novabase.pt* <http://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 <joel.silva at novabase.pt>
> > *www.novabase.pt* <http://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 <joel.silva at novabase.pt>
> > > *www.novabase.pt* <http://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.
> > >  ------------------------------
> > > Save all your chat conversations. Find them online.
> > > <http://in.rd.yahoo.com/tagline_webmessenger_3/*http://in.messenger.yahoo.com/webmessengerpromo.php>
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> >
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> >
>
>
>
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