[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.





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