You'll want some rules in your sip.conf to handle the connection from
SER. A
starting point might be:
[<ser ip addr>:<ser port ?= 5060>]
type=peer
context=<my sip context name>
tos=lowdelay ; tos delay
allow=ulaw ; dtmfmode=inband only works with ulaw
or alaw!
dtmfmode=inband ; Choices are inband, rfc2833, or info
You'll then want some rules in extensions.conf to accept the call and
redirect it
to mailboxes defined in your voicemail.conf or in MySQL. Something like:
[general]
context=<my sip context name>
switch => Realtime/<my sip context name>@extensions
static=yes
[<my sip context name>]
exten => _uXXXXX,1,VoiceMail(${EXTEN}@<my sip context name>)
exten => _XXXXX,1,VoiceMail(${EXTEN}@<my sip context name>)
exten => _bXXXXX,1,VoiceMail(${EXTEN}@<my sip context name>))
exten => #,2,Hangup ; Hang them up.
Steve
harry gaillac wrote:
Hello,
I try set Ua---SER----Asterisk (voicemail/ARA)
|
Ua
ser stable
asterisk cvs head
I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to asterisk
voicemail in failure route.
How may I configure extensions.conf and ser.cfg ?
I have been trying without success!
Regards
Harry
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