[SIPForum-discussion] Unified voicemail

Sun Dec 16 18:38:32 UTC 2007

Hi Joel,

Please find my commens through your lines;hopefully comments are useful.

selamlar,
halit

Date: Mon, 10 Dec 2007 16:10:42 +0000From: joel.silva at novabase.ptTo: rohin.koul at gmail.com; discussion at sipforum.orgSubject: Re: [SIPForum-discussion] Unified voicemail

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.
-Lets say A: pstn, B: sip. 
-Here of course you will need SIP-AS to generate an INV for voicemail which has 
Req uri: voicemail at hedehodo.com.
-Think there are two sides SIP and PSTN or SIP and GSM;
for translating the sip messages you will need 
pstn or gsm gateway (as you mentioned)
When the message is recorded, the user hang-up and the message is sent to the user e-mail.
The message recording will be handled by voicemail server.
Recording and generating an email.
So what we need are basicly;
*pstn call handling component
*pstn gateway for communicating sip proxy
*B has CRVM(call forwarding to voicemail) so SIP-AS to serve.
*voicemail server to establish a session with A
What are the components that you see that I need to implement such a scenario? Please consider a non-IMS network.
Thanks,

Joel SilvaAnalyst
.............................................................................................................
NovabaseAv. Eng. Duarte Pacheco, 15F . 1099-078 Lisboa - PortugalTel. (+351) 213 836 300 . Fax (+351) 213 836 301 .mailto:joel.silva at novabase.ptwww.novabase.pt

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Rohin KoulSent: segunda-feira, 10 de Dezembro de 2007 13:15To: discussion at sipforum.orgSubject: 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 SilvaAnalyst
.............................................................................................................
NovabaseAv. Eng. Duarte Pacheco, 15F . 1099-078 Lisboa - PortugalTel. (+351) 213 836 300 . Fax (+351) 213 836  301 .mailto:joel.silva at novabase.ptwww.novabase.pt

From: discussion-bounces at sipforum.org [mailto: discussion-bounces at sipforum.org] On Behalf Of Rohin KoulSent: segunda-feira, 10 de Dezembro de 2007 12:06To: discussion at sipforum.orgSubject: 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 SilvaAnalyst
.............................................................................................................
NovabaseAv. Eng. Duarte Pacheco, 15F . 1099-078 Lisboa - PortugalTel. (+351) 213 836 300 . Fax (+351) 213 836  301 .mailto:joel.silva at novabase.ptwww.novabase.pt

From: discussion-bounces at sipforum.org [mailto: discussion-bounces at sipforum.org] On Behalf Of Ritambhara MittalSent: segunda-feira, 10 de Dezembro de 2007 10:33To: Ritambhara Mittal; nidhi jain; discussion at sipforum.orgSubject: [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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