[Asterisk-Users] X-Lite and Voicemail

[Rennes Neps]

Hello!

You can try playing with dtmf settings in sip.conf  For example all 
Grandstream phones in my box use dtmfmode=info and Snom phones use 
dtmfmode=rfc2833
I got the same behaviour with snom's when dtmfmode was wrong. Hope it helps.

[202]
type=friend
auth=md5
md5secret=secret
nat=1
qualify=500
host=dynamic
canreinvite=no
callerid="" <202>
context=class1
;codec=g729
dtmfmode=info
pickupgroup=9
callgroup=9
mailbox=202 at class1

Regards
Rennes Neps




Joe Hetrick wrote:

> I've been doing a little battle with asterisk voicemail and X-Lite 
> (under OSX).
>
> I can leave voicemail without a problem, everything seems to run into 
> problems upon checking VM.
>
> The log is pretty descriptive:
>
>     -- Executing Ringing("SIP/xlite2-70d1", "") in new stack
>     -- Executing Wait("SIP/xlite2-70d1", "2") in new stack
>     -- Executing VoiceMailMain("SIP/xlite2-70d1", "") in new stack
>     -- Playing 'vm-login' (language 'en')
>     -- Playing 'vm-password' (language 'en')
>     -- Incorrect password '55555555555' for user '22222222222' 
> (context = <any>)
>     -- Playing 'vm-incorrect-mailbox' (language 'en')
>
>
> Now, in this case, I would have entered "2" (just once) when prompted 
> for the mailbox.  Again, I'd have entered the password (which in this 
> case is 5555).  As you can see, There are more than enough 2's and 
> 5's, which cause the user/pass check to fail for VM.
>
> To verify that this was the issue, I let the user entry timeout, which 
> causes the extension's CID info (as entered in sip.conf) to be 
> utilized and set the password to be 55555555555, and I was in.
>
> Pretty standard, simple extensions.conf, sip.conf, voicemail.conf 
> entries in the sample config files:
>
> sip.conf
> [xlite2]
> type=friend
> ;regexten=5678               ; When they register, create extension 1234
> username=xlite2
> secret=password
> mailbox=2 at softphone
> ; auth=md5
> callerid="Joe Laptop" <2>
> host=dynamic
> ;nat=yes                       ; X-Lite is behind a NAT router
> ;canreinvite=no                ; Typically set to NO if behind NAT
> context=softphone              ; our extensions for softphones
> disallow=all
> ;allow=gsm                     ; GSM consumes far less bandwidth than 
> ulaw
> allow=ulaw
> allow=alaw
>
> extensions.conf
> # Voicemail Checking
> exten => 50,1,Ringing
> exten => 50,2,Wait,2
> exten => 50,3,VoicemailMain
> exten => 50,4,Hangup
>
> voicemail.conf
> 2 => 5555,xlite2
>
>     
> I've just begun fiddling with asterisk, is there something I've 
> overlooked?
>
> Thanks!
>
> Joe
>