RE : [Asterisk-Users] Voicemail + SIP Message header
Title: RE : [Asterisk-Users] Voicemail + SIP Message header
Hi Deepak,
I had a similar setup, However I was able to configure the softswitch to send a prefix followed by the number dialled in the to field. This way I could route the call to the right mailbox and be able to play the right busy or away message.
Can you say what softswitch you are using ?
Umar
I am trying to use Asterisk as a "pure" voicemail system and have the following
setup:
I have the * setup as a SIP peer to a softswitch. When someone calls a number on the softswitch and no one picks up the phone, the softswitch forwards the call to the * using SIP. The message header of the SIP INVITE has the number originally called in the "To:" field, but the INVITE is still being sent to the number asterisk is configured for.
Is there any way that I can configure asterisk to "read" the To: field in the message header of the SIP INVITE and then go to the mailbox of the corresponding number?
Thanks
Deepak
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RE: [Asterisk-Users] Voicemail + SIP Message header
Lal, Deepak (Contractor) [EMAIL PROTECTED] wrote:
> I am trying to use Asterisk as a "pure" voicemail system and have the
> following setup: I have the * setup as a SIP peer to a softswitch. When
> someone calls a number on the softswitch and no one picks up the phone,
> the softswitch forwards the call to the * using SIP. The message header
> of the SIP INVITE has the number originally called in the "To:" field,
> but the INVITE is still being sent to the number asterisk is configured
> for.
>
> Is there any way that I can configure asterisk to "read" the To: field in
> the message header of the SIP INVITE and then go to the mailbox of the
> corresponding number?
>
It sounds to me as if you're forwarding all VM calls to a single
extension on the Asterisk box, such as 1000, and are then trying to
work out which mailbox the call should be sent to, with no further IDs
to use as a guide.
If you're only using Asterisk as an answering machine (a bit of a
waste, in my view) then you could forward all calls to individual
extensions on the Asterisk box, so extension "2101" on your switch
would defer to "[EMAIL PROTECTED]" for VM.
Once you have that, you could capture all incoming calls with a single
context in "extensions.conf", such as the following:
[]
exten => _,1,Answer
exten => _,2,Wait(1)
exten => _,3,VoiceMail2(su${EXTEN})
exten => _,4,Hangup
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED]
_/ _/ _/_/_/_/ _/_/_/_/ _/_/
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Re: [Asterisk-Users] Voicemail + SIP Message header
The ${RDNIS} variable in the dialplan would contain that information.
${RDNIS} for SIP is in CVS HEAD. A patch for 0.7.2 is at
http://www.fnords.org/~eric/asterisk/downloads/
On Thu, 2004-03-25 at 14:07, Lal, Deepak (Contractor) wrote:
> I am trying to use Asterisk as a "pure" voicemail system and have the following
> setup:
> I have the * setup as a SIP peer to a softswitch. When someone calls a number on
> the softswitch and no one picks up the phone, the softswitch forwards the call
> to the * using SIP. The message header of the SIP INVITE has the number
> originally called in the "To:" field, but the INVITE is still being sent to the
> number asterisk is configured for.
>
> Is there any way that I can configure asterisk to "read" the To: field in the
> message header of the SIP INVITE and then go to the mailbox of the corresponding
> number?
>
> Thanks
>
> Deepak
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upgrades can NOT be deducted as a gambling loss."
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Re: [Asterisk-Users] Voicemail + SIP Message header
Lal, Deepak (Contractor) wrote: I am trying to use Asterisk as a "pure" voicemail system and have the following setup: I have the * setup as a SIP peer to a softswitch. When someone calls a number on the softswitch and no one picks up the phone, the softswitch forwards the call to the * using SIP. The message header of the SIP INVITE has the number originally called in the "To:" field, but the INVITE is still being sent to the number asterisk is configured for. Is there any way that I can configure asterisk to "read" the To: field in the message header of the SIP INVITE and then go to the mailbox of the corresponding number? So all INVITES go to the same URI, regardless of the called number? Is it impossible to change that? If it is, one could implement a SIPTO variable, but I can't see a general need for that. Already have a SIPFROM variable in chan_sip2.c (hint,hint). /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
