[Asterisk-Users] Xfering a call
Rich Adamson
radamson at routers.com
Wed Jan 12 16:46:02 MST 2005
> I'm having an issue when I transfer a call to another SIP extension it sees
> that the sip phone is not there and goes to voicemail but in my case it
> transfers to the main voicemail instead of the users voicemail.
>
> Here is what my SIP extensions look like in the extension.conf file
>
> exten => 3957,1,Dial(${Theresa},20,Tt)
> exten => 3957,2,VoicemailMain2(u${TheresaVM})
> exten => 3957,3,Hangup
> exten => 3957,102,VoicemailMain2(b${TheresaVM})
> exten => 3957,103,Hangup
Change the above from VoicemailMain2 to Voicemail and it will work
as expected.
The 3,Hangup isn't required... remove it. 103 isn't actually needed
either.
More information about the asterisk-users
mailing list