Thankyou, this was a great primer for me also.
Chris
trixter aka Bret McDanel wrote:
On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote:
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to these dids
has to be routed to the right handset thru Asterisk, no Ivr at this time, at
least an answering machine in case of busy or not available users.
For the rest, we need to be able to have external calls to pstn, or even to
other sip phones form other providers.
Is that enough?
Not for 100% setup, but enoughto at least get you started. From what I
understand this is what it appears you want (I may be wrong, if I am let
me know).
You will want voicemail for each user. This is configured in
voicemail.conf
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
You will need to edit sip.conf for the voip provider (register and
context) and if the office workers use sip to asterisk one for each of
them as well.
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
Lastly you will want to create a dialplan so that when a call comes in
from the DID it will then dial the appropriate user and if busy/no
answer goto voicemail. This is done from extensions.conf.
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
You may want a macro like:
[macro-dialvmb]
exten => s,1,Dial(${ARG1},20,t)
exten => s,2,Voicemail(u${ARG2})
exten => s,3,Hangup
exten => s,102,Voicemail(b${ARG2})
exten => s,103,Hangup
Then for each inbound DID something like:
exten => 18005551212,1,Macro(dialvmb,SIP/user1,1234)
where user1 is the user defined in sip.conf, 1234 is the voicemail
extension defined in voicemail.conf and 18005551212 is the extension
that a given did goes to (ie last part of the register line).
Hope this helps
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