[asterisk-users] How can i send my sip channel 3 to mailbox 2? Please Help!
Jonathan GF
jonathangf at gmail.com
Sun Sep 2 09:03:51 CDT 2007
Hi folks,
i'm trying to configure my extensions.conf as small as posible and for
that reason i'm using macros. The problem is that maybe I have a
misunderstood the concept for the directive "mailbox" in sip.conf.
Under my knowledge configuring the mailbox directive to the mailbox I
want would be enought to leave an retreive messages in that voicemail
box. Of course it seems to be that i was wrong :/
What i'm trying is to have ONLY 2 voicemail boxes and depending which
extensions i'm dialing send the caller to one or the other, but not
send based on the called id/name, but to that mailbox i want (mailbox
1 or mailbox 2, just this).
The error i'm getting is:
WARNING[2102]: app_voicemail.c:2461 leave_voicemail: No entry in
voicemail config file for '3'
The error is correct... i don't have a voicemail box named/numbered
"3" but this is the behavior i want to control. How can i send my sip
channel 3 to mailbox 2?
I'm a bit stuck and would appreciate so much your help.
The block that is causing me headache is that:
-------------------------------- SIP.CONF --------------------------------
[3]
context = internal
type = friend
username = 3
secret = pwd3
callerid = "Studio" <3>
host = dynamic
nat = no
mailbox = 2
qualify = yes
canreinvite = no
callgroup = 2
pickupgroup = 2,1
dtmfmode = rfc2833
---------------------- EXTENSIONS.CONF -----------------------------
[internal]
exten => _x,1,Macro(diallocal|${EXTEN}|SIP/${EXTEN}|15)
[macro-diallocal]
exten => s,1,Dial(${ARG2}|${ARG3}|Tr)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${ARG1})
exten => s-NOANSWER,2,Hangup()
exten => s-BUSY,1,Voicemail(b${ARG1})
exten => s-BUSY,2,Hangup
exten => _s-.,1,Hangup
------------------------- VOICEMAIL.CONF ----------------------------------
[zonemessages]
europe=Europe/Madrid|'vm-received' Q 'digits/at' R
[default]
1 => 1,Main Phone,,,saycid=yes|delete=no|tz=europe
2 => 2,The Studio,,,saycid=yes|delete=no|tz=europe
All configuration (sip, extensions, voicemail, etc...) is available @
http://www.surestorm.com/asterisk/ for those that want to help.
Thanks in advance.
Best regards,
Jonathan GF
More information about the asterisk-users
mailing list