[SIPForum-discussion] Unified voicemail

Mon Dec 10 16:10:42 UTC 2007

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 <mailto: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 <mailto:joel.silva at novabase.pt> 
	www.novabase.pt <http://www.novabase.pt/> 

________________________________

	From: discussion-bounces at sipforum.org [mailto:
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
<mailto:joel.silva at novabase.pt> 
		www.novabase.pt <http://www.novabase.pt/> 

________________________________

		From: discussion-bounces at sipforum.org [mailto:
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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