Hi,
I encounter problems to setup my VoiceMail(Asterisk SVN-trunk-r48389) since I have up to date Asterisk from SVN.
sip.conf
[code][100]
type=friend
secret=100
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=100@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/100
context=internal
canreinvite=no
callerid=Vincent <100>
accountcode=100
[101]
type=friend
secret=101
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
mailbox=101@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/101
context=internal
canreinvite=no
callerid=Murielle <101>
accountcode=101[/code]
extensions.conf
[internal]
exten => 100,1,Dial(SIP/100&IAX2/103,15,r)
exten => 100,n,VoiceMail(u100@default)
exten => 100,102,VoiceMail(b100@default)
exten => 101,1,Dial(SIP/101,15,r)
exten => 101,n,VoiceMail(u101@default)
exten => 101,102,VoiceMail(b101@default)
exten => 123,1,VoiceMailMain()
exten => 222,1,Directory(default,internal,f)
exten => 900,1,MeetMe(900)
include => Free-Telecom-OutGoing
voicemail.conf
[default]
100 => 100,xxx,xxx@free.fr,envelope=no
101 => 101,yyy,yyy@free.fr,envelope=no
And my logs when trying to leave a message :
--- Executing [101@internal:1] Dial("SIP/100-081cb508", "SIP/101|15|r") in new stack
Really destroying SIP dialog '6a936c3d03f818bc691b8f395765212b@127.0.0.1' Method: INVITE
[Dec 11 22:25:13] WARNING[15010]: app_dial.c:1289 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [101@internal:2] VoiceMail("SIP/100-081cb508", "u101@default") in new stack
[Dec 11 22:25:13] WARNING[15010]: app_voicemail.c:2863 leave_voicemail: No entry in voicemail config file for 'u101'
Curiously, the same configurations worked with a previous asterisk build.
Any idea ?
Thanks for advance.
